The trace I posted earlier is what I see with tcpdump when attempting a
call. There is no other INVITE shown in the trace:
http://pastebin.com/raw/C4iymTbh

The trace seems to end abruptly in the middle of the SDP, so I think it
could be due to packet fragmentation.
On 6 May 2016 4:18 pm, "Bogdan-Andrei Iancu" <[email protected]> wrote:

> So that meas the INVITE never gets to the callee ?? maybe it is not
> properly routed .
>
> Do you see (with ngrep or tcpdump) the INVITE being sent out by opensips
> towards callee ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 06.05.2016 12:56, Nabeel wrote:
>
> Hi,
>
> Thanks for the idea about packet compression. By 'call fails to connect',
> I meant the call does not connect to the callee, ie. the callee's phone
> does not ring after the INVITE (despite using TURN server).
>
> This was a public WiFi network and that was all I could get at the time. I
> am using OpenSIPS version 2.1.
>
> Nabeel
> On 6 May 2016 9:16 am, "Bogdan-Andrei Iancu" <[email protected]> wrote:
>
>> Hi,
>>
>> Hard to analyze a call based on the INVITE packet only :). Still the SIP
>> signaling does not show any ALG interference (also not sure if the capture
>> was done before or after the ALG). Also, what you mean by "call fails" ?no
>> reply, negative reply , no audio ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 05.05.2016 22:35, Nabeel wrote:
>>
>>
>> Please check the following SIP trace taken within a WiFi network. The
>> call fails to connect despite the INVITE request and using a non-standard
>> port. Could this be caused by SIP ALG, or some unopened RTP port on the
>> router?
>>
>> http://pastebin.com/raw/C4iymTbh
>>
>>
>> _______________________________________________
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>>
>>
>>
>
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