[asterisk-bugs] [Asterisk 0012902]: Video RTP is not sended to originating SIP extension when using IAX2 to interconnect both servers

[noreply at bugs.digium.com]

The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=12902 
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Reported By:                albersag
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12902
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             06-20-2008 02:46 CDT
Last Modified:              06-20-2008 02:46 CDT
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Summary:                    Video RTP is not sended to originating SIP extension
when using IAX2 to interconnect both servers
Description: 
When i use IAX2 to interconnect two servers, Asterisk does not send RTP in
H.263,263+ or H.264 to the originating SIP extension of the call.

It´s always reproducible and it always happens in originating extension.
Other party could see you but you could not see him/her.

Analyzing RTP/SIP captures, i see Asterisk1 (from where i originate SIP
Video Call) does not receive RTP packets in video códec sended by client
SIP2 in other Asterisk2

It does not depend on Softhphone or Hardphone used. We had same problem
with Grandstream GXV3000, or instead Eyebeam Softphone.
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Issue History 
Date Modified   Username       Field                    Change               
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06-20-08 02:46  albersag       Asterisk Version          => 1.4.18          
06-20-08 02:46  albersag       SVN Branch (only for SVN checkou => N/A          
  
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