[Freeswitch-dev] SIP Timeout waiting for 200 ACK

[freeswitch at dalethatcher.com]

I'm seeing an issue during a bridge where the '200' from freeswitch is
not being ACK'd and Freeswitch is correctly dropping the connection.
However I belive that the Contact header in the 200 packet is causing
the problem.  Am I missing a config parameter?

If I back out the revision 5208 change to mod_sofia.c from the trunk
source then the problem goes away. (SIPTAG_CONTACT_STR(tech_pvt->to_uri)
back to SIPTAG_CONTACT_STR(tech_pvt->profile->url)).

My test setup is (Freeswitch is behind a NAT):

  External SIP -> Freeswitch -> Gizmo

The freeswitch config for the extension is:

  <extension name="test">
    <condition field="destination_number" expression="^test$">
      <action application="bridge" data="sofia/gizmo/17470355893 at proxy01.sipphone.com"/>
    </condition>
  </extension>

I've put a full copy of the log here:

  http://pastebin.freeswitch.org/3008

My internet facing profile is:

  <profile name="internet">
    <settings>
      <param name="debug" value="1"/>
      <param name="rfc2833-pt" value="101"/>
      <param name="sip-port" value="5060"/>
      <param name="dialplan" value="XML"/>
      <param name="dtmf-duration" value="100"/>
      <param name="codec-prefs" value="$${global_codec_prefs}"/>
      <param name="codec-ms" value="20"/>
      <param name="use-rtp-timer" value="true"/>
      <param name="rtp-timer-name" value="soft"/>
      <param name="rtp-ip" value="$${bind_server_ip}"/>
      <param name="sip-ip" value="$${bind_server_ip}"/>
      <param name="accept-blind-reg" value="true"/>
      <param name="nonce-ttl" value="60"/>
      <param name="ext-rtp-ip" value="80.189.98.58"/>
      <param name="ext-sip-ip" value="80.189.98.58"/>
      <param name="sip-domain" value="sip.dalethatcher.com"/>
    </settings>
  </profile>

With the current trunk the SIP packet that does not get a response is:

  send 1182 bytes to udp/[193.111.201.32]:5060 at 13:05:15.136189:
    ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 193.111.201.32;branch=z9hG4bK614f.db2cc426.0
   Via: SIP/2.0/UDP 193.111.200.182;branch=z9hG4bK614f.9dbef354.0
   Via: SIP/2.0/UDP 194.145.189.10:3670;branch=z9hG4bK00E0F556495005A857E00000230F
   Record-Route: <sip:193.111.201.32;lr=on;ftag=00E0F556495005A857E00000222A>
   Record-Route: <sip:193.111.200.182;lr=on;ftag=00E0F556495005A857E00000222A>
   From: <sip:02085555129 at 194.145.189.10>;tag=00E0F556495005A857E00000222A
   To: <sip:442070968842 at 193.111.200.182>;tag=r5ae93vm8aK1p
   Call-ID: 00E0F556495005A857E000001549 at 194.145.189.10
   CSeq: 35816 INVITE
   Contact: <sip:442070968842 at 193.111.200.182:5060>
   User-Agent: FreeSWITCH(mod_sofia)
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: 100rel, precondition
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 290
   
   v=0
   o=FreeSWITCH 6489556414011969222 9057524047013517960 IN IP4 192.168.93.5
   s=FreeSWITCH
   c=IN IP4 80.189.98.58
   t=0 0
   a=sendrecv
   m=audio 10002 RTP/AVP 0 96 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 telephone-event/8000
   a=fmtp:96 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   m=image 0 UDPTL 9
    ------------------------------------------------------------------------

Backing out the change modifies the Contact header to:

  Contact: <sip:mod_sofia at 80.189.98.58:5060>

Which gets an ACK packet and doesn't drop the call.

thanks,

- Dale