[Freeswitch-dev] SIP Timeout waiting for 200 ACK
Brian West
brian.west at mac.com
Tue May 29 11:54:15 EDT 2007
I think tony put up a fix for this can you try latest trunk
/b
On May 29, 2007, at 10:12 AM, Mike Murdock wrote:
> I am dealing with the same problem... Brian, there is something
> wrong here. The effect is my calls get terminated every two minutes
> +hen my sipura re-registers. I have a trace with full debug
> captured if you want to see.
>
> Michael Murdock
> VP Operations & Development
> Switchmaxx Product Line
> 125 N. Market. Suite 1520
> Wichita KS. 67202
> (316) 932-2693
>
> -----Original Message-----
> From: Brian West [mailto:brian.west at mac.com]
> Sent: Tuesday, May 29, 2007 10:14 AM Eastern Standard Time
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] SIP Timeout waiting for 200 ACK
>
> 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
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
> freeswitch-dev
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20070529/c8740f67/attachment-0001.html
More information about the Freeswitch-dev
mailing list