[Freeswitch-users] INCOMPATIBLE_DESTINATION

Jonas Gauffin jonas.gauffin at gmail.com
Fri Aug 27 04:36:13 PDT 2010


Excellent information thanks. The server is a dedicated voip server and
nothing else is running on it (and therefore the port should be free).

I've talked to my voip provider and they were kind enough to give me their
traces.

Freeswitch get's their SDP on the bleg (their software have modified the
trace a bit):

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 212.247.E.E;rport=5060;branch=z9hG4bK52XcXtpQBp3BH
From: "blablabla" <sip:02345678 at 212.247.E.E>;tag=rccD75cXD61Kr
To: <sip:0771221122 at voipprovider.se <sip%3A0771221122 at voipprovider.se>
>;tag=686577264
Call-ID: c3e8e0e8-2c3c-122e-479c-1fc6e9408ca4
CSeq: 1106961 INVITE
Contact: <sip:0771221122 at 212.151.Y.Y:5060>
Record-Route: <sip:130.244.Z.Z;lr;ftag=rccD75cXD61Kr;t2diag=c82.4127173>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
Content-Type: application/sdp
Content-Length: 169

v=0
o=- 46541649 0 IN IP4 130.244.x.x
s=Cisco SDP 0
c=IN IP4 130.244.x.x
t=0 0
m=audio 18928 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

which makes FS cancel the call leg:

CANCEL sip:0771221122 at voipprovider.se <sip%3A0771221122 at voipprovider.se>SIP/2.0
Via: SIP/2.0/UDP 212.247.E.E;rport;branch=z9hG4bK52XcXtpQBp3BH
Max-Forwards: 68
From: "blablabla" <sip:02345678 at 212.247.E.E>;tag=rccD75cXD61Kr
To: <sip:0771221122 at voipprovider.se <sip%3A0771221122 at voipprovider.se>>
Call-ID: c3e8e0e8-2c3c-122e-479c-1fc6e9408ca4
CSeq: 1106961 CANCEL
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0



On Fri, Aug 27, 2010 at 9:57 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Yes, G.729a and G.729b are incorrect and your device is at fault... only
> other phones of the same type would probably recognise it without issue.
>
> PCMA should be used in this case though.
>
> RTP payload numbers are spread through multiple RFCs. Anything in the
> 96-127 range is dynamic and the codec is determined from the matching rtpmap
> line, any of the static numbers don't need a rtpmap line to work. IANA
> oversees assignment of the static numbers and they have the full list:
> http://www.iana.org/assignments/rtp-parameters
>
> As you can see 0=PCMU, 8=PCMA, and G.729 should use 18. Support of annex B
> is specified in the fmtp parameter, not the codec name - e.g. "a=fmtp:18
> annexb=no". Annex A never needs to be specified as it can be read normally
> by plain G.729, so it's just up to the implementation on whether it wants to
> save quality or cpu when encoding.
>
> Do you have any other applications running which would also be using the
> RTP port range? A call will fail if it tries to use a port that's already in
> use, perhaps with that message. FS should avoid using ports it's already
> using, but can't know about any other programs on the system.
>
> -Steve
>
>
>
> On 27 August 2010 08:03, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
>
>> It doesn't happen every time and it's on a production system with a bit of
>> volume. therefore a bit hard to get SIP traces. I'll try if Anthony really
>> needs them.
>>
>> FS do say this:
>> 2010-08-27 07:14:10.758750 [DEBUG] switch_ivr_originate.c:3111sofia/external/
>> 0700123456 at 212.151.Y.Y:5060 Media Establishment Failed
>>
>> In which RFC are codec names defined? rfc3551 defines "G729" but no
>> "G.729A" or "G.729B". But as you say, shouldn't FS use PCMA in any case?
>>
>>
>>
>> On Fri, Aug 27, 2010 at 8:43 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Ordinarily it'd mean there was a problem with the codecs (e.g. needing to
>>> use an unsupported codec or transcode a codec that only works as a
>>> passthrough one).
>>>
>>> Looks like it should have gone through with PCMA (8) though. Can you
>>> repeat the call with sip trace on? Perhaps the incompatible destination
>>> comes from an endpoint.
>>>
>>> 'sofia profile <profilename> siptrace on' from the CLI, replace on with
>>> off to turn it off again.
>>>
>>> -Steve
>>>
>>>
>>>
>>>   On 27 August 2010 07:26, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>>
>>>>   Hello,
>>>>
>>>> Why do this call result in INCOMPATIBLE_DESTINATION?
>>>>
>>>> http://pastebin.freeswitch.org/13736
>>>>
>>>> Regards,
>>>>   Jonas
>>>>
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>>>
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>>
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