[asterisk-users] Asterisk 11 WebSockets.

James Mortensen james.mortensen at a-cti.com
Tue Sep 4 15:52:06 CDT 2012

qasimakhan <at> gmail.com <qasimakhan <at> gmail.com> writes:

> Hi,I was testing with newly introduced websocket support in asterisk 11. I 
have successfully implemented everything except when i try to make a call i get 
no audio. I have tried both SipML5 as well as SIP-JS as clients. the call get 
connected but i never hear any audio stream. I however get the following warning
> WARNING[2626][C-00000000]: chan_sip.c:9686 process_sdp: Ignoring video stream 
offer because port number is zero
> When i turn rtp debug on i can see RTP getting through. 
> CLI Output:        http://pastebin.pk/16sip.conf:            
http://pastebin.pk/17http.conf:           http://pastebin.pk/19extensions.conf: 
> --
> _____________________________________________________________________

According to the Asterisk developers, this is an issue in the hands of the 
browser developers. Here is the wiki page on the Asterisk 11 SIP over 

At this time, no media is flowing.


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