[Freeswitch-dev] SIP Timeout waiting for 200 ACK

[Brian West]

I think that since we are a B2BUA tech_pvt->profile->url is the right  
way.  The other way is correct if you're a proxy.  Remember  
FreeSWITCH NOT a proxy.

/b

On May 29, 2007, at 8:58 AM, freeswitch at dalethatcher.com wrote:

> 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
>
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