[Freeswitch-dev] SIP Timeout waiting for 200 ACK
freeswitch at dalethatcher.com
freeswitch at dalethatcher.com
Wed May 30 04:54:59 EDT 2007
Updating to revision 5229 fixed the issue I was seeing.
thanks for the help,
- Dale
On Tue, May 29, 2007 at 10:54:15AM -0500, Brian West wrote:
> 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
> >>
> >> _______________________________________________
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> >> Freeswitch-dev at lists.freeswitch.org
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> >
> >
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