[asterisk-users] Asterisk 11 WebSockets.

[qasimakhan at gmail.com]

Thanks :).


Regards,
Qasim

On Wed, Sep 5, 2012 at 1:52 AM, James Mortensen
<james.mortensen at a-cti.com>wrote:

> 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
>
>
> --
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