[Freeswitch-users] Issue with Invites without SDP

Wed Nov 10 07:50:13 PST 2010

And I really should read emails properly rather than jumping to
conclusions... you've already tried that. :o)

Try posting full debug-level logs for the entire length of the call.
Context is important, not just the CRIT error but what happened
leading up to it. That'll also probably show the HTTP request problem.
What're you using - mod_xml_curl?

My hunch is it's something about codec negotiation since you're
setting bypass media, late negotiation and disabling transcoding.

Try removing the disable-transcoding parameter. The effect of that is
to only offer to the B-leg the codec from the A-Leg. Since the A-Leg
has no codec selected because there's no SDP to offer one and you're
delaying selecting a codec (late negotiation) then there's perhaps no
codec to offer to the B-leg.

-Steve

On 10 November 2010 15:41, Steven Ayre <steveayre at gmail.com> wrote:
> The default behaviour for INVITEs without SDP is to reject the call
> because media cannot be negotiated.
>
> It's possible to accept these INVITEs by setting the following
> parameter on the SIP profile:
>
> <param name="enable-3pcc" value="true"/>
>
> See these previous discussions from the archive:
> http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg03504.html
> http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg11825.html
>
>
> Warm regards,
> -Steve
>
>
>
> On 10 November 2010 15:22, SIC FS LIST <sicfslist at gmail.com> wrote:
>> Hello,
>> I have a working FS instance I am using as a redirect server (to serve up
>> LNP requests).  It works fine on calls with invites that have SDP and does
>> not work with invites without SDP.  I enabled 3pcc to true thinking that
>> would fix the issue.  Version info is FreeSWITCH Version 1.0.6
>> (hacked-20100921T052029Z).
>> With the console log level set to debug the only thing I see is this message
>> (just before returning a 480):
>> freeswitch at lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT]
>> switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout
>> waiting for next instruction in CS_NEW!
>> I also do not ever see FS to an xml http req (which is how we control the
>> dialplan).
>> freeswitch at lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at
>> 14:57:50.934431:
>> ------------------------------------------------------------------------
>> INVITE sip:19033226103 at X.X.X.X:5060;user=phone SIP/2.0
>> Via: SIP/2.0/UDP
>> X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5
>> Max-Forwards: 70
>> Contact: <sip:X.X.X.X:5060>
>> To: <sip:19033226103 at X.X.X.X:5060>
>> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001
>> Call-ID: 993186-34928005397-80464 at ens.com
>> CSeq: 693501 INVITE
>> Date: Tue, 09 Nov 2010 15:29:25 GMT
>> Supported: 100rel
>> User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152
>> Content-Length: 0
>> ------------------------------------------------------------------------
>> send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324:
>> ------------------------------------------------------------------------
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5
>> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001
>> To: <sip:19033226103 at X.X.X.X:5060>
>> Call-ID: 993186-34928005397-80464 at ens.com
>> CSeq: 693501 INVITE
>> User-Agent: lnpdal0001.sipinterchange.com
>> Content-Length: 0
>> ------------------------------------------------------------------------
>> send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429:
>> ------------------------------------------------------------------------
>> SIP/2.0 480 Temporarily Unavailable
>> Via: SIP/2.0/UDP
>> X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5
>> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001
>> To: <sip:19033226103 at X.X.X.X:5060>;tag=t7ycB64mX33mQ
>> Call-ID: 993186-34928005397-80464 at ens.com
>> CSeq: 693501 INVITE
>> User-Agent: lnpdal0001.sipinterchange.com
>> Accept: application/sdp
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, refer
>> Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE"
>> Content-Length: 0
>> Remote-Party-ID: "19033226103"
>> <sip:19033226103 at X.X.X.X>;party=calling;privacy=off;screen=no
>> ------------------------------------------------------------------------
>> Here is the sofia profile:
>> <profile name="external">
>> <aliases>
>> <alias name="outbound"/>
>> <alias name="nat"/> <!-- for backwards compatibility -->
>> </aliases>
>> <domains>
>> <domain name="all" alias="false" parse="true"/>
>> </domains>
>> <settings>
>> <param name="user-agent-string" value="lnpdal0001"/>
>> <param name="debug" value="0"/>
>> <param name="sip-trace" value="no"/>
>> <param name="context" value="public"/>
>> <param name="enable-100rel" value="false"/>
>> <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="$${global_codec_prefs}"/>
>> <param name="use-rtp-timer" value="true"/>
>> <param name="rtp-timer-name" value="soft"/>
>> <param name="rtp-ip" value="X.X.X.X"/>
>> <param name="sip-ip" value="X.X.X.X"/>
>> <param name="manage-presence" value="false"/>
>> <param name="inbound-codec-negotiation" value="generous"/>
>> <param name="bind-params" value="transport=udp"/>
>> <param name="tls" value="$${external_ssl_enable}"/>
>> <!-- <param name="pass-rfc2833" value="true"/> -->
>> <!-- <param name="inbound-proxy-media" value="true"/> -->
>> <param name="inbound-bypass-media" value="true"/>
>> <param name="inbound-late-negotiation" value="true"/>
>> <param name="accept-blind-reg" value="false"/>
>> <param name="accept-blind-auth" value="true"/>
>> <param name="nonce-ttl" value="60"/>
>> <param name="disable-transcoding" value="true"/>
>> <param name="auth-calls" value="false"/>
>> <param name="inbound-reg-force-matching-username" value="false"/>
>> <param name="auth-all-packets" value="false"/>
>> <param name="rtp-timeout-sec" value="300"/>
>> <param name="rtp-hold-timeout-sec" value="1800"/>
>> <param name="challenge-realm" value="auto_to"/>
>> <param name="enable-3pcc" value="true"/>
>> </settings>
>> </profile>
>> Any thoughts / ideas / help would be greatly appreciated.
>> Thanks!
>> SDR
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>>
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