[asterisk-users] Asterisk 11 WebSockets.

[James Mortensen]

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: 
http://pastebin.pk/20Regards,Qasim
> 
> 
> --
> _____________________________________________________________________

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 
WebSockets:  
https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support

At this time, no media is flowing.

James