Thanks :).<br><br><br>Regards,<br>Qasim<br><br><div class="gmail_quote">On Wed, Sep 5, 2012 at 1:52 AM, James Mortensen <span dir="ltr"><<a href="mailto:james.mortensen@a-cti.com" target="_blank">james.mortensen@a-cti.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">qasimakhan <at> <a href="http://gmail.com" target="_blank">gmail.com</a> <qasimakhan <at> <a href="http://gmail.com" target="_blank">gmail.com</a>> writes:<br>
<br>
><br>
><br>
> Hi,I was testing with newly introduced websocket support in asterisk 11. I<br>
<div><div class="h5">have successfully implemented everything except when i try to make a call i get<br>
no audio. I have tried both SipML5 as well as SIP-JS as clients. the call get<br>
connected but i never hear any audio stream. I however get the following warning<br>
><br>
> WARNING[2626][C-00000000]: chan_sip.c:9686 process_sdp: Ignoring video stream<br>
offer because port number is zero<br>
><br>
><br>
> When i turn rtp debug on i can see RTP getting through.<br>
><br>
> CLI Output: <a href="http://pastebin.pk/16sip.conf" target="_blank">http://pastebin.pk/16sip.conf</a>: <br>
<a href="http://pastebin.pk/17http.conf" target="_blank">http://pastebin.pk/17http.conf</a>: <a href="http://pastebin.pk/19extensions.conf" target="_blank">http://pastebin.pk/19extensions.conf</a>:<br>
<a href="http://pastebin.pk/20Regards,Qasim" target="_blank">http://pastebin.pk/20Regards,Qasim</a><br>
><br>
><br>
</div></div>> --<br>
> _____________________________________________________________________<br>
<br>
According to the Asterisk developers, this is an issue in the hands of the<br>
browser developers. Here is the wiki page on the Asterisk 11 SIP over<br>
WebSockets:<br>
<a href="https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support" target="_blank">https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support</a><br>
<br>
At this time, no media is flowing.<br>
<br>
James<br>
<br>
<br>
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