Asterisk 11.2.2 / videocalls [solved]

Elleni · Veteran Joined: 23 May 2006 Posts: 1270

I have successfully installed asterisk and am playing with it and freepbx on a testsystem with some sip phones. Until now - with asterisk 1.8 everything worked perfectly but since update to version 11.2 I cannot videocall anymore, and in asterisk console I get message: chan_sip.c:10136 process_sdp: Ignoring video stream offer because port number is zero.

Anyone an idea? Should I fill a bug @bugs.gentoo.org?

I reverted back to 1.8.20 and on a call with two softphones with same app (Join) videocalls do work, but not when trying to call from Join softphone to Jitsi (another softphone).

After going back to asterisk 11.2.2, videocalls do not work anymore, not even from Join to Join. It gives me following message in asterisk cli:

Ignoring video stream offer because port number is zero

When trying to activate camera while beeing on an audio call (Join-Join), it says: translate.c:343 framein: no samples for gsmtolin.

But as said, it works without any probs on asterisk 1.8.20 with Join softphone to Join Softphone, installed on two iPhones.
Here are sip show peer xxxx for 11.2.2 version of asterisk:

http://pastebin.com/QhtnBjNt
http://pastebin.com/zUYZxVHu

and following a sip show peer xxxx for asterisk 1.8.20 where videocall is working from one sipclient (Join) to another sipclient (Join):
http://pastebin.com/ThP412wT

Please let me know, if I should provide some further infos, and/or if I should file a bug.

I had already asked in irc asterisk channel, but they said, go deal with your siphone software

jkroon · Posted: Wed Jul 03, 2013 9:56 am Post subject:

Hi,

We missed each other on IRC but I got your question, here was my response:

<jkroon> Elleni,
<jkroon> that looks more like a core asterisk bug, so filing a bug on bugs.gentoo.org won't help much.
<jkroon> a more helpful output would be CLI output with sip set debug on

Don't know what the official policy is, but I'd rather have a long paste inside "code" blocks here than having to reference some pastebin URL.

Using video myself, and I explicitly set to only allow h264. That might make a difference for you, failing that, full output from asterisk CLI with:

Elleni · Veteran Joined: 23 May 2006 Posts: 1270

Ok, I tried asterisk-11.4 and there videocalling is working again, so I guess, I can put solved on this topic, although when having asked for this problem in asterisk irc channel, they were like: go deal with your siphone software... :roll:

If I have some time left in the evening I will revert back to 11.2 to paste a full output of debug enabled cli.