[Freeswitch-users] INCOMPATIBLE_DESTINATION

Discussion:

[Freeswitch-users] INCOMPATIBLE_DESTINATION - did FS not offer a codec?

Avi Marcus

2011-09-13 15:01:05 UTC

I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't
show PCMU in it - did FS not offer any codecs?
FS log: http://pastebin.freeswitch.org/17316
PCAP: http://ge.tt/95tFji7

Can someone tell me what's going on here?

-Avi
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Kristian Kielhofner

2011-09-13 15:08:57 UTC

Permalink

Try setting verbose SDP:

http://wiki.freeswitch.org/wiki/Variable_verbose_sdp

I see a failed bridge to my second carrier.?The??sip_local_sdp_str doesn't
show PCMU in it - did FS not offer any codecs?
FS log:?http://pastebin.freeswitch.org/17316
PCAP:?http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
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--
Kristian Kielhofner

Michael Collins

2011-09-13 15:28:49 UTC

Permalink

For the record, line #107 contains the codecs (and other stuff) that FS
offered:

m=audio 16480 RTP/AVP 8 0 9 101 13

The "8 0 9 101 13" represent the codecs (and other stuff). What do those
numbers mean? This wiki page has the list:

http://wiki.freeswitch.org/wiki/RTP_payload_list

The SDP spec says that for non-dynamic payload numbers you are NOT required
to have the "a=rtpmap" lines. This makes sense because those payload numbers
don't change meaning until you get up to 96 and higher. The issue is that
some devices are broken and require the "a=rtpmap" lines for the payload
types below number 96. The verbose_sdp chan var mentioned by KK tells FS to
add all those a=rtpmap lines for the non-dynamic payload types. You should
only set this var when absolutely necessary because those verbose SDP's can
cause the packets to be larger than the allowable MTU size.

Dontcha just *love* SIP/SDP interop? :)

-MC

Post by Avi Marcus
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't
show PCMU in it - did FS not offer any codecs?
FS log: http://pastebin.freeswitch.org/17316
PCAP: http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
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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Avi Marcus

2011-09-13 23:53:42 UTC

Permalink

OK then!
Looking at my call logs, it seems to mostly happen with one particular
carrier. Can I set verbose_sdp=true in the bridge string? e.g.
"[verbose_sdp=true]sofia/gateway/... "?

Thanks,
-Avi

Post by Michael Collins
For the record, line #107 contains the codecs (and other stuff) that FS
m=audio 16480 RTP/AVP 8 0 9 101 13
The "8 0 9 101 13" represent the codecs (and other stuff). What do those
http://wiki.freeswitch.org/wiki/RTP_payload_list
The SDP spec says that for non-dynamic payload numbers you are NOT required
to have the "a=rtpmap" lines. This makes sense because those payload numbers
don't change meaning until you get up to 96 and higher. The issue is that
some devices are broken and require the "a=rtpmap" lines for the payload
types below number 96. The verbose_sdp chan var mentioned by KK tells FS to
add all those a=rtpmap lines for the non-dynamic payload types. You should
only set this var when absolutely necessary because those verbose SDP's can
cause the packets to be larger than the allowable MTU size.
Dontcha just *love* SIP/SDP interop? :)
-MC

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Michael Collins

2011-09-14 00:15:00 UTC

Permalink

I believe that is exactly the way to use it.
-MC

Post by Avi Marcus
OK then!
Looking at my call logs, it seems to mostly happen with one particular
carrier. Can I set verbose_sdp=true in the bridge string? e.g.
"[verbose_sdp=true]sofia/gateway/... "?
Thanks,
-Avi

Post by Michael Collins
For the record, line #107 contains the codecs (and other stuff) that FS
m=audio 16480 RTP/AVP 8 0 9 101 13
The "8 0 9 101 13" represent the codecs (and other stuff). What do those
http://wiki.freeswitch.org/wiki/RTP_payload_list
The SDP spec says that for non-dynamic payload numbers you are NOT
required to have the "a=rtpmap" lines. This makes sense because those
payload numbers don't change meaning until you get up to 96 and higher. The
issue is that some devices are broken and require the "a=rtpmap" lines for
the payload types below number 96. The verbose_sdp chan var mentioned by KK
tells FS to add all those a=rtpmap lines for the non-dynamic payload types.
You should only set this var when absolutely necessary because those verbose
SDP's can cause the packets to be larger than the allowable MTU size.
Dontcha just *love* SIP/SDP interop? :)
-MC

Post by Avi Marcus
I see a failed bridge to my second carrier. The sip_local_sdp_str
doesn't show PCMU in it - did FS not offer any codecs?
FS log: http://pastebin.freeswitch.org/17316
PCAP: http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
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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Steven Ayre

2011-09-14 07:47:02 UTC

Permalink

You should also be able to set it in the gateway configuration, I believe:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Variables

If that works then that'll probably make it far easier to manage as you
won't need to put in any logic to alter the dialstring.

-Steve

Post by Michael Collins
For the record, line #107 contains the codecs (and other stuff) that FS
m=audio 16480 RTP/AVP 8 0 9 101 13
The "8 0 9 101 13" represent the codecs (and other stuff). What do those
http://wiki.freeswitch.org/wiki/RTP_payload_list
The SDP spec says that for non-dynamic payload numbers you are NOT
required to have the "a=rtpmap" lines. This makes sense because those
payload numbers don't change meaning until you get up to 96 and higher. The
issue is that some devices are broken and require the "a=rtpmap" lines for
the payload types below number 96. The verbose_sdp chan var mentioned by KK
tells FS to add all those a=rtpmap lines for the non-dynamic payload types.
You should only set this var when absolutely necessary because those verbose
SDP's can cause the packets to be larger than the allowable MTU size.
Dontcha just *love* SIP/SDP interop? :)
-MC

Post by Avi Marcus
I see a failed bridge to my second carrier. The sip_local_sdp_str
doesn't show PCMU in it - did FS not offer any codecs?
FS log: http://pastebin.freeswitch.org/17316
PCAP: http://ge.tt/95tFji7
Can someone tell me what's going on here?
-Avi
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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