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Asterisk 11.2.2 / videocalls [solved]
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Elleni
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Joined: 23 May 2006
Posts: 220

PostPosted: Fri Jun 28, 2013 8:12 am    Post subject: Asterisk 11.2.2 / videocalls [solved] Reply with quote

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 :(


Last edited by Elleni on Wed Jul 03, 2013 1:01 pm; edited 5 times in total
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jkroon
Tux's lil' helper
Tux's lil' helper


Joined: 15 Oct 2003
Posts: 108
Location: South Africa

PostPosted: Wed Jul 03, 2013 9:56 am    Post subject: Reply with quote

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:

Code:
core set verbose 10
sip set debug on


Will pretty much be the only thing that would enable myself or chainsaw to help you. Neither of us hangs on the forums much, so please do prod us on IRC.
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Elleni
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Joined: 23 May 2006
Posts: 220

PostPosted: Wed Jul 03, 2013 1:00 pm    Post subject: Reply with quote

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.
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