Re: [Linphone-developers] linphonec 3.4.0 fails to initiate SIP calls

[Pedro Sanchez]

On 02/11/2011 09:25 PM, Pedro Sanchez wrote:
> Hello,
>
> I downloaded linphonec 3.4.0 and cross-compiled it for ARM. I had to
> assume OSIP_SUCCESS == 0 for the compilation to succeed according to
> this e-mail:
>
> http://lists.nongnu.org/archive/html/linphone-developers/2011-02/msg00061.html
>
>
> Linphone successfully receives calls, audio and everything else works,
> but I can't get it to initiate calls. This is what I get:
>
> linphonec> call sip:address@hidden
> Establishing call id to <sip:address@hidden>, assigned id 1
> linphonec> Contacting <sip:address@hidden>
> linphonec> Call 1 to <sip:address@hidden> in progress.
> Remote ringing.
> linphonec> Call 1 to <sip:address@hidden> ringing.
> Warning: Remote end <sip:address@hidden> seems to have
> disconnected, the call is going to be closed.
> linphonec> Call ended
> linphonec> Call 1 with <sip:address@hidden> ended.
> ortp-error-Could not remove the call from the list !!!
> Call 1 with <sip:address@hidden> error.
>
> A SIP trace shows the INVITE from linphone followed by the TRYING and
> RINGING from the remote end. This is followed almost immediately by a
> CANCEL from linphonec.
>
> I must say that once in a blue moon it works but I can't tell what
> exactly is necessary for this to work. For sure it doesn't work on the
> very first call attempt. And as I keep trying to call I notice that the
> call id numbers are always increasing, is this expected?
>
> Also, when I exit I get this:
>
> linphonec> quit
> Terminating...
> *** glibc detected *** linphonec: munmap_chunk(): invalid pointer:
> 0x00038bc0 ***
> Aborted
>
> I was running linphonec 3.3.2 before on the same host, calling the same
> peer, and linphonec never failed. Any pointers on what to do would be
> appreciated.
>
> Thank you,

The problem was a patch that I forgot to mention which aims at making 
the RTP timeout event that ends a call on media inactivity work from the 
beginning of the SIP negotiation and not from the moment the first RTP 
packet is received.

I was given a patch that was discussed somewhere else which indeed 
implements the feature but unfortunately breaks the initiation of new 
calls. I attach a modified patch that fixes the problem in case someone 
else is interested.

--
Pedro

0001-Make-RTP-traffic-timer-start-when-SIP-negotiation-st.patch
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