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

From: Pedro Sanchez
Subject: Re: [Linphone-developers] linphonec 3.4.0 fails to initiate SIP calls
Date: Sat, 12 Feb 2011 12:14:34 -0500
User-agent: Mozilla/5.0 (X11; U; Linux x86_64; en-US; rv: Gecko/20101208 Lightning/1.0b2 Thunderbird/3.1.7

On 02/11/2011 09:25 PM, Pedro Sanchez wrote:

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:

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
*** glibc detected *** linphonec: munmap_chunk(): invalid pointer:
0x00038bc0 ***

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

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.


Attachment: 0001-Make-RTP-traffic-timer-start-when-SIP-negotiation-st.patch
Description: Text Data

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