[Freeswitch-users] Wrong port on response

Discussion:

David Aldworth

2008-09-19 23:08:29 UTC

Hello -

Got an issue with Freeswitch not responding on the port that the
initial request was made on. I'm not beyond believing that it is a NAT
or router issue except that I can register a Cisco phone from another
location or a softphone from the same location without any problem.
This Aastra just won't work for some reason.

We have connectile-dysfuntion turned on. Otherwise we are using the
default profile settings. Auth is on (as you can see from the below).
Basically, the Reg request comes from port 41450, but freeswitch
responds on port 5060. Again, other UA's work fine, just one Cisco and
one Aastra from this site do not. Meanwhile a soft phone from this
site, and the same model cisco from another site do not work.

SIP dump and external profile are below. Thank you for any help. David

U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060
REGISTER sip:atl.teliax.net SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab.
Max-Forwards: 70.
Content-Length: 0.
To: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>.
From: Aastra Test
<sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6.
Call-ID: ecae0e043ee65c5c5340749a9d28235f at 192.168.1.192.
CSeq: 288881481 REGISTER.

;expires=300.

Allow-Events: talk,hold,conference.
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO.
Expires: 300.
User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/
v3.2.8.45.
.
U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34.
From: Aastra Test
<sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6.
To: Aastra Test
<sip:pleasehelp(aastra)@atl.teliax.net>;tag=Sarvm1DjmU3Zg.
Call-ID: ecae0e043ee65c5c5340749a9d28235f at 192.168.1.192.
CSeq: 288881481 REGISTER.
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: 100rel, timer, precondition, path, replaces.
WWW-Authenticate: Digest realm="atl.teliax.net", nonce="d3538e86-9d86-
dd11-82bf-001143e64915", algorithm=MD5, qop="auth".
Content-Length: 0.


<profile name="external">

<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>

<domains>
<domain name="$${domain}" parse="true"/>
</domains>

<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>
</profile>
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Brian West

2008-09-19 23:27:37 UTC

Permalink

Post by David Aldworth
Hello -
Got an issue with Freeswitch not responding on the port that the
initial request was made on. I'm not beyond believing that it is a
NAT or router issue except that I can register a Cisco phone from
another location or a softphone from the same location without any
problem. This Aastra just won't work for some reason.
We have connectile-dysfuntion turned on. Otherwise we are using the
default profile settings. Auth is on (as you can see from the
below). Basically, the Reg request comes from port 41450, but
freeswitch responds on port 5060. Again, other UA's work fine, just
one Cisco and one Aastra from this site do not. Meanwhile a soft
phone from this site, and the same model cisco from another site do
not work.
SIP dump and external profile are below. Thank you for any help. David

This isn't a bug. If you notice the phone explicitly said in its
contact for us to contact them via 192.168.1.192:5060 so you'll need
to enable stun on the phone or rport. If you were to enable rport on
the aastra it would just work correctly. If Aastra doesn't support
RFC3581 or STUN then they are worthless phones just like Polycom. Its
not the registrars problem to fix your nat issues. The phone should
support RFC3581 (rport) or STUN and it would just work like the Snom's
do.

Try adding this param to your sofia profile. It will break cisco
phones or any other phone that follows the sip spec. This explicitly
breaks RFC to accommodate broken phones.

<param name="NDLB-force-rport" value="true"/> in your sofia profile.

/b

Post by David Aldworth
U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060
REGISTER sip:atl.teliax.net SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab.
Max-Forwards: 70.
Content-Length: 0.
From: Aastra Test
Call-ID: ecae0e043ee65c5c5340749a9d28235f at 192.168.1.192.
CSeq: 288881481 REGISTER.
Contact: Aastra Test
Allow-Events: talk,hold,conference.
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO.
Expires: 300.
User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/
v3.2.8.45.
.
U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34.
From: Aastra Test
To: Aastra Test
Call-ID: ecae0e043ee65c5c5340749a9d28235f at 192.168.1.192.
CSeq: 288881481 REGISTER.
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: 100rel, timer, precondition, path, replaces.
WWW-Authenticate: Digest realm="atl.teliax.net",
nonce="d3538e86-9d86-dd11-82bf-001143e64915", algorithm=MD5,
qop="auth".
Content-Length: 0.

<profile name="external">

<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>
<domains>
<domain name="$${domain}" parse="true"/>
</domains>
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>
</profile>
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Brian West

2008-09-19 23:34:37 UTC

Permalink

Yep just checked in my 55i that I have and no STUN nor RPORT. Great
for the LAN silly for the WAN. the force-rport is your only option
and its a global per profile setting.

/b

Post by David Aldworth
User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/
v3.2.8.45.

Darren Schreiber

2008-09-21 16:45:47 UTC

Permalink

The Aastra 55i should have STUN support, but not on the firmwares released
with the phone initially. Check your firmware version here:

http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21
669.htm

For the 480i, though, I think you're SOL.

- Darren

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org]
Sent: Friday, September 19, 2008 4:35 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Wrong port on response

Yep just checked in my 55i that I have and no STUN nor RPORT. Great for the
LAN silly for the WAN. the force-rport is your only option and its a global
per profile setting.

/b

Post by David Aldworth
User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/
v3.2.8.45.

Brian West

2008-09-21 16:51:38 UTC

Permalink

I have looked all over the web interface and can't find it. I have
the latest firmware too... weird. Still no rport support..?!?

/b

Post by Darren Schreiber
The Aastra 55i should have STUN support, but not on the firmwares released
http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21
669.htm
For the 480i, though, I think you're SOL.
- Darren

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unknown

2008-09-21 19:03:51 UTC

Permalink

Hi Brian, I have Aastra 55i, it does have STUN and Rport support, and I like
its rport support. On web interface, under network settings->Advanced
Network Settings->Rport (RFC 3581)Enabled for Rport
or STUN Server STUN Port
for STUN support, however I like rport better, which I can have call
forwarding working seamlessly for freeswitch in public ip address, even my
Aastra is behind my doule-NATed home network.

Best regards,
Chris

I have looked all over the web interface and can't find it. I have the
latest firmware too... weird. Still no rport support..?!?
/b
The Aastra 55i should have STUN support, but not on the firmwares released
http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21
669.htm
For the 480i, though, I think you're SOL.
- Darren
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Brian West

2008-09-21 22:14:42 UTC

Permalink

That would def. solve the problem :P

/b

Post by unknown
Hi Brian, I have Aastra 55i, it does have STUN and Rport support,
and I like its rport support. On web interface, under network
settings->Advanced Network Settings->Rport (RFC 3581)Enabled for Rport
or
STUN Server
STUN Port
for STUN support, however I like rport better, which I can have call
forwarding working seamlessly for freeswitch in public ip address,
even my Aastra is behind my doule-NATed home network.
Best regards,
Chris

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David Aldworth

2008-10-15 19:11:44 UTC

Permalink

Brian,

How do you go about configuring the 7960 to FreeSWITCH? I've got the
same problem where the Contact field in the SIP header is set to port
5060 but the NAT is assigning an alternative port. There doesn't seem
to be any stun or rport support on the phones. I can set the external
IP, just not the port. So when FS responds it always goes to the wrong
place.

Thanks,
David

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