Discussion:
[Freeswitch-users] SDP With T.38 in INVITE Problem
ivdreg ivdreg
2010-02-19 10:42:27 UTC
Permalink
Hi all,

Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.

2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065]
10.10.1.100:17058->10.10.1.110:0codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR:
[Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING


Thanks
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ivdreg ivdreg
2010-02-22 12:48:57 UTC
Permalink
Hi All,

Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.

Regards.

2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
[Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
Thanks
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Michael Jerris
2010-02-22 14:45:08 UTC
Permalink
If the endpoint does not correctly follow the sdp o/a model its not going to work. This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do. What is the device?

Mike
Post by ivdreg ivdreg
Hi All,
Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
m=audio 26850 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=image 0 udptl 19
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
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ivdreg ivdreg
2010-02-22 15:11:26 UTC
Permalink
Hi Michael,

This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this
is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM
ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.

Thanks

2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not going
to work. This is not a "problem" with the sofia library, this is intended
behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
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Michael Jerris
2010-02-22 15:29:22 UTC
Permalink
if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.

Mike
Post by ivdreg ivdreg
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW
(this is ITSP equipment and I don't know vendor) to our SIP subscribers with
ATCOM ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not going
to work. This is not a "problem" with the sofia library, this is intended
behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
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ivdreg ivdreg
2010-02-22 16:49:07 UTC
Permalink
Hi Michael,

As I said in a previous mails I know exactly what is happening.
In working setup:

ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.

I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
FreeSwitch for some reasons. The problem is:

INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP:
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:
transferredTCF

And reply 200 OK contains in SDP:
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20

Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in
SDP:
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

In this case everything works fine. Line *m=image 0 udptl 19 *is removed by
YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
0 udptl 19" *call brakes as you can see in my first mail.

I don't want to compare or discus YATE and FS functionality or something
else. I just see difference in behavior and because I'm not a big expert
don't know witch implementation is more accurate according standards. And
second: Is it impossible for me to upgrade all CPE so only thing I can do is
to fix it on server side. That is because I ask for a help.


Thanks to all.


2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.
Mike
Post by ivdreg ivdreg
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW
(this is ITSP equipment and I don't know vendor) to our SIP subscribers with
ATCOM ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not
going to work. This is not a "problem" with the sofia library, this is
intended behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
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Steve Underwood
2010-02-22 17:10:40 UTC
Permalink
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.

Steve
ivdreg ivdreg
2010-02-22 17:28:43 UTC
Permalink
It comes form subscriber (ATCOM CPEs). As you know OpenSIPS is just a proxy
so cannot generate or rewrite SDP (generally speaking). Yes, I cannot fix
CPEs only server :(

Regards

2010/2/22 Steve Underwood <steveu at coppice.org>
Post by ivdreg ivdreg
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.
Steve
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Michael Jerris
2010-02-22 17:29:22 UTC
Permalink
the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't support
it.
Post by ivdreg ivdreg
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.
Steve
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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ivdreg ivdreg
2010-02-22 20:23:00 UTC
Permalink
Thanks Michael,

I will try to find some solution/workaround.

2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't
support it.
Post by ivdreg ivdreg
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.
Steve
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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ivdreg ivdreg
2010-02-22 20:23:00 UTC
Permalink
Thanks Michael,

I will try to find some solution/workaround.

2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't
support it.
Post by ivdreg ivdreg
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.
Steve
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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ivdreg ivdreg
2010-02-22 17:28:43 UTC
Permalink
It comes form subscriber (ATCOM CPEs). As you know OpenSIPS is just a proxy
so cannot generate or rewrite SDP (generally speaking). Yes, I cannot fix
CPEs only server :(

Regards

2010/2/22 Steve Underwood <steveu at coppice.org>
Post by ivdreg ivdreg
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.
Steve
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-------------- next part --------------
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Michael Jerris
2010-02-22 17:29:22 UTC
Permalink
the port 0 with PT of 19 is sofia rejecting the sdp becuase we don't support
it.
Post by ivdreg ivdreg
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.
Steve
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Michael Jerris
2010-02-24 08:32:15 UTC
Permalink
if you want clarity on this, read the rfc for sdp offer answer. You are not supposed to remove an m= line in an answer, if something is doing that, it is incorrect.

Mike
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
m=audio 34788 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=audio 16330 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=image 0 udptl 19
In this case everything works fine. Line m=image 0 udptl 19 is removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) "m=image 0 udptl 19" call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or something else. I just see difference in behavior and because I'm not a big expert don't know witch implementation is more accurate according standards. And second: Is it impossible for me to upgrade all CPE so only thing I can do is to fix it on server side. That is because I ask for a help.
Thanks to all.
2010/2/22 Michael Jerris <mike at jerris.com>
if you want to see what is going on, crank up the debug in freeswitch and sofia and you should see exactly what is going on.
Mike
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM ATA and IP Phone. We use now in production YATE for terminating and originating GWs to ITSPs and FS as main routing logic (backend). We want to switch YATE to FS for a GW also but we faced this problem. This not happens if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
If the endpoint does not correctly follow the sdp o/a model its not going to work. This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do. What is the device?
Mike
Post by ivdreg ivdreg
Hi All,
Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
m=audio 26850 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=image 0 udptl 19
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
_______________________________________________
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ivdreg ivdreg
2010-02-24 14:52:21 UTC
Permalink
Hi Michael,

Thanks for clarifying. Unfortunately we don't live in prefect world. I was
fixed that by disabling T.38 in codec negotiation and everything works fine.

Thanks Again.

2010/2/24 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
if you want clarity on this, read the rfc for sdp offer answer. You are
not supposed to remove an m= line in an answer, if something is doing that,
it is incorrect.
Mike
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is removed
by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or something
else. I just see difference in behavior and because I'm not a big expert
don't know witch implementation is more accurate according standards. And
second: Is it impossible for me to upgrade all CPE so only thing I can do is
to fix it on server side. That is because I ask for a help.
Thanks to all.
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.
Mike
Post by ivdreg ivdreg
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW
(this is ITSP equipment and I don't know vendor) to our SIP subscribers with
ATCOM ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not
going to work. This is not a "problem" with the sofia library, this is
intended behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185
Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
http://lists.freeswitch.org/mailman/options/freeswitch-users
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-------------- next part --------------
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ivdreg ivdreg
2010-02-24 14:52:21 UTC
Permalink
Hi Michael,

Thanks for clarifying. Unfortunately we don't live in prefect world. I was
fixed that by disabling T.38 in codec negotiation and everything works fine.

Thanks Again.

2010/2/24 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
if you want clarity on this, read the rfc for sdp offer answer. You are
not supposed to remove an m= line in an answer, if something is doing that,
it is incorrect.
Mike
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is removed
by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or something
else. I just see difference in behavior and because I'm not a big expert
don't know witch implementation is more accurate according standards. And
second: Is it impossible for me to upgrade all CPE so only thing I can do is
to fix it on server side. That is because I ask for a help.
Thanks to all.
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.
Mike
Post by ivdreg ivdreg
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW
(this is ITSP equipment and I don't know vendor) to our SIP subscribers with
ATCOM ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not
going to work. This is not a "problem" with the sofia library, this is
intended behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185
Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
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Steve Underwood
2010-02-22 17:10:40 UTC
Permalink
Hi Michael,
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing
server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS --->
Subscriber.
I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop)
INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE
between FreeSwitch (routing server) and YATE (GW - SIP Interop)
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
In this case everything works fine. Line *m=image 0 udptl 19 *is
removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop)
*"m=image 0 udptl 19" *call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or
something else. I just see difference in behavior and because I'm not
a big expert don't know witch implementation is more accurate
according standards. And second: Is it impossible for me to upgrade
all CPE so only thing I can do is to fix it on server side. That is
because I ask for a help.
You said the the broken line "m=image 0 udptl 19" goes from OpenSIPS to
YATE. Do you know if it originates from the OpenSIPS box or the
subscriber? If it originates from the OpenSIPS box it should be reported
to them. If its from the subscriber, well...... your chances of getting
anything fixed are usually small.

Steve
Michael Jerris
2010-02-24 08:32:15 UTC
Permalink
if you want clarity on this, read the rfc for sdp offer answer. You are not supposed to remove an m= line in an answer, if something is doing that, it is incorrect.

Mike
Post by ivdreg ivdreg
Hi Michael,
As I said in a previous mails I know exactly what is happening.
ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl) ---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
transferredTCF
m=audio 34788 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=audio 16330 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=image 0 udptl 19
In this case everything works fine. Line m=image 0 udptl 19 is removed by YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) "m=image 0 udptl 19" call brakes as you can see in my first mail.
I don't want to compare or discus YATE and FS functionality or something else. I just see difference in behavior and because I'm not a big expert don't know witch implementation is more accurate according standards. And second: Is it impossible for me to upgrade all CPE so only thing I can do is to fix it on server side. That is because I ask for a help.
Thanks to all.
2010/2/22 Michael Jerris <mike at jerris.com>
if you want to see what is going on, crank up the debug in freeswitch and sofia and you should see exactly what is going on.
Mike
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM ATA and IP Phone. We use now in production YATE for terminating and originating GWs to ITSPs and FS as main routing logic (backend). We want to switch YATE to FS for a GW also but we faced this problem. This not happens if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
If the endpoint does not correctly follow the sdp o/a model its not going to work. This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do. What is the device?
Mike
Post by ivdreg ivdreg
Hi All,
Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
m=audio 26850 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=image 0 udptl 19
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
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ivdreg ivdreg
2010-02-22 16:49:07 UTC
Permalink
Hi Michael,

As I said in a previous mails I know exactly what is happening.
In working setup:

ITSP ---> YATE (GW - Frontend) ---> FreeSwitch (routing server/xml_curl)
---> YATE (GW - SIP Interop) ---> OpenSIPS ---> Subscriber.

I prefer to change YATE (GW - Frontend) and YATE (GW - SIP Interop) with
FreeSwitch for some reasons. The problem is:

INVITE comes from ITSP and goes to subscriber via OpenSIPS. INIVITE between
FreeSwitch (routing server) and YATE (GW - SIP Interop) contains SDP:
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:
transferredTCF

And reply 200 OK contains in SDP:
*m=audio 34788 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20

Reply 200 OK SDP between YATE (GW - SIP Interop) and OpenSIPS contains in
SDP:
*m=audio 16330 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

In this case everything works fine. Line *m=image 0 udptl 19 *is removed by
YATE.
But same test with FreeSWITCH on the place of YATE (GW - SIP Interop) *"m=image
0 udptl 19" *call brakes as you can see in my first mail.

I don't want to compare or discus YATE and FS functionality or something
else. I just see difference in behavior and because I'm not a big expert
don't know witch implementation is more accurate according standards. And
second: Is it impossible for me to upgrade all CPE so only thing I can do is
to fix it on server side. That is because I ask for a help.


Thanks to all.


2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.
Mike
Post by ivdreg ivdreg
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW
(this is ITSP equipment and I don't know vendor) to our SIP subscribers with
ATCOM ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not
going to work. This is not a "problem" with the sofia library, this is
intended behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send
signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-------------- next part --------------
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Michael Jerris
2010-02-22 15:29:22 UTC
Permalink
if you want to see what is going on, crank up the debug in freeswitch and
sofia and you should see exactly what is going on.

Mike
Post by ivdreg ivdreg
Hi Michael,
This happens when ONLY IF initial INVITE is coming with T.38 from a GW
(this is ITSP equipment and I don't know vendor) to our SIP subscribers with
ATCOM ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.
Thanks
2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not going
to work. This is not a "problem" with the sofia library, this is intended
behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
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ivdreg ivdreg
2010-02-22 15:11:26 UTC
Permalink
Hi Michael,

This happens when ONLY IF initial INVITE is coming with T.38 from a GW (this
is ITSP equipment and I don't know vendor) to our SIP subscribers with ATCOM
ATA and IP Phone. We use now in production YATE for terminating and
originating GWs to ITSPs and FS as main routing logic (backend). We want to
switch YATE to FS for a GW also but we faced this problem. This not happens
if initial INVITE nave no T.38 offered and later re-INVITE with T.38 with
valid SDP port.

Thanks

2010/2/22 Michael Jerris <mike at jerris.com>
Post by Michael Jerris
If the endpoint does not correctly follow the sdp o/a model its not going
to work. This is not a "problem" with the sofia library, this is intended
behavior and what we are supposed to do. What is the device?
Mike
Hi All,
Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS
ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA]
[DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
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Michael Jerris
2010-02-22 14:45:08 UTC
Permalink
If the endpoint does not correctly follow the sdp o/a model its not going to work. This is not a "problem" with the sofia library, this is intended behavior and what we are supposed to do. What is the device?

Mike
Post by ivdreg ivdreg
Hi All,
Actually while seeking the solution in internet I see some people having this problem with sofia library. I'm not sure that SIP reply in this case contains a valid SDP (I think that teminating endpoint is broken) but in my opinion if we have at least one valid media type in SDP (video, audio, image ...) call must be established. Can someone comment and/or help me with this issue.
Regards.
2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway FreeSwitch drops the call because of media error ? As I see from log only T.38 port is zero and SDP has also media port. Is it possible to configure FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
m=audio 26850 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
m=image 0 udptl 19
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP [sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->10.10.1.110:0 codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR: [Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46 sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP -> CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590 (sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
-------------- next part --------------
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ivdreg ivdreg
2010-02-19 10:42:27 UTC
Permalink
Hi all,

Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.

2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF

2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4135 Remote SDP:
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*

2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065]
10.10.1.100:17058->10.10.1.110:0codec: 0 ms: 20
2010-02-19 12:26:21.255201 [ERR] sofia_glue.c:2879 AUDIO RTP REPORTS ERROR:
[Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 Standard HANGUP, cause:
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING


Thanks
-------------- next part --------------
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ivdreg ivdreg
2010-02-22 12:48:57 UTC
Permalink
Hi All,

Actually while seeking the solution in internet I see some people having
this problem with sofia library. I'm not sure that SIP reply in this case
contains a valid SDP (I think that teminating endpoint is broken) but in my
opinion if we have at least one valid media type in SDP (video, audio, image
...) call must be established. Can someone comment and/or help me with this
issue.

Regards.

2010/2/19 ivdreg ivdreg <ivdreg at gmail.com>
Post by ivdreg ivdreg
Hi all,
Dose someone have a problem that if there T.38 in coming from gateway
FreeSwitch drops the call because of media error ? As I see from log only
T.38 port is zero and SDP has also media port. Is it possible to configure
FS to do not break a call but if media is OK.
2010-02-19 12:26:14.305244 [NOTICE] switch_channel.c:666 New Channel
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065[6cd9f634-411d-df11-99ca-003048bb99cc]
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:3369 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_NEW -> CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_INIT
2010-02-19 12:26:14.305244 [DEBUG] switch_core_state_machine.c:338
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State INIT
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:83 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 SOFIA INIT
2010-02-19 12:26:14.305244 [DEBUG] sofia_glue.c:1377 sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 Patched SDP
---
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 21108 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.110
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 21108 udptl t38
c=IN IP4 10.10.1.110
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
+++
v=0
o=- 3779949069 5166785777234550622 IN IP4 206.132.232.212
s=session
t=0 0
m=audio 17058 RTP/AVP 18 4 8 0
c=IN IP4 10.10.1.100
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=image 17058 udptl t38
c=IN IP4 10.10.1.100
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
2010-02-19 12:26:14.305244 [DEBUG] mod_sofia.c:117 (sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065) State Change CS_INIT -> CS_ROUTING
......
v=0
o=FreeSWITCH 1266548331 1266548332 IN IP4 10.10.1.110
s=FreeSWITCH
c=IN IP4 10.10.1.110
t=0 0
*m=audio 26850 RTP/AVP 8*
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
*m=image 0 udptl 19*
2010-02-19 12:26:21.255201 [DEBUG] sofia.c:4124 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 entering state [ready][200]
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2285 Send signal
sofia/backend/YYYYYYYYYY at 10.10.1.110 [BREAK]
2010-02-19 12:26:21.255201 [NOTICE] sofia.c:4668 Channel [sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065] has been answered
2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2288 Set Codec
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 PROXY/8000 20 ms 160 samples
*2010-02-19 12:26:21.255201 [DEBUG] sofia_glue.c:2571 PROXY AUDIO RTP
[sofia/backend/XXXXXXXXXX at 10.10.1.110:7065] 10.10.1.100:17058->
10.10.1.110:0 codec: 0 ms: 20
[Missing remote port]
2010-02-19 12:26:21.255201 [NOTICE] sofia_glue.c:2880 Hangup sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]*
2010-02-19 12:26:21.255201 [DEBUG] switch_channel.c:2063 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [KILL]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change CS_HANGUP
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:411 Channel sofia/backend/
XXXXXXXXXX at 10.10.1.110:7065 hanging up, cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:46
DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:499
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State HANGUP going to sleep
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:333
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State Change CS_HANGUP ->
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_core_session.c:1018 Send signal
sofia/backend/XXXXXXXXXX at 10.10.1.110:7065 [BREAK]
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:314
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) Running State Change
CS_REPORTING
2010-02-19 12:26:21.255201 [DEBUG] switch_ivr_originate.c:3185 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2010-02-19 12:26:21.255201 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: DESTINATION_OUT_OF_ORDER
2010-02-19 12:26:21.255201 [DEBUG] switch_core_state_machine.c:590
(sofia/backend/XXXXXXXXXX at 10.10.1.110:7065) State REPORTING
Thanks
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