<div dir="ltr"><div class="gmail_default" style="font-size:small;display:inline">Is video call currently supported for WebRTC? My experience with the demo site is that after a few seconds , the video frame freezes while audio continues to work. Is this the expected behavior for now?</div>
<div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51);display:inline"><br></div></div><div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51);display:inline">
Thanks,</div></div><div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51);display:inline"><br></div></div><div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51);display:inline">
Henry</div></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Jul 5, 2013 at 3:15 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">It is the right thing to do to turn on dtls. As far as how it should be handled in jsssip, you will have to talk to them about that. I don't think we plan at this point to support webrtc without dtls as everything I have seen says it will be required by the browsers at some point anyways.<div>
<br></div><div>Mike</div><div><div class="h5"><div><br><div><div>On Jul 5, 2013, at 6:05 PM, Henry Huang <<a href="mailto:red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a>> wrote:</div><br><blockquote type="cite">
<div dir="ltr"><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">I confirm that the changes made by Iwan worked and gave me audio. Thanks, Iwan. </div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">
<br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">But now the question is that is it a hack on the js side or is it the right thing to do? And is it going to be merged into the JsSIP core?</div>
<div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">Thanks,</div>
<div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small;color:rgb(51,51,51)">Henry</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Jul 5, 2013 at 2:27 PM, Iwan Budi Kusnanto <span dir="ltr"><<a href="mailto:ibk@labhijau.net" target="_blank">ibk@labhijau.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Henry,<br>
I can make it jssip demo works by modify this line<br>
<br>
this.peerConnection = new<br>
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);<br>
<br>
into<br>
<br>
constraints["optional"] = [];<br>
constraints["optional"].push({'DtlsSrtpKeyAgreement': 'true'});<br>
<br>
this.peerConnection = new<br>
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);<br>
<br>
On Sat, Jul 6, 2013 at 1:12 AM, Anthony Minessale<br>
<div><div><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>> wrote:<br>
> Maybe because we hacked in dtls support?<br>
><br>
><br>
><br>
> On Fri, Jul 5, 2013 at 12:58 PM, Henry Huang <<a href="mailto:red.rain.seven@gmail.com" target="_blank">red.rain.seven@gmail.com</a>><br>
> wrote:<br>
>><br>
>> I think I have found the issue. If I use the JsSIP website demo to<br>
>> register to <a href="http://webrtc.freeswitch.org/" target="_blank">webrtc.freeswitch.org</a> then I will be able to replicate the no<br>
>> audio issue on <a href="http://webrtc.freeswitch.org/" target="_blank">webrtc.freeswitch.org</a><br>
>><br>
>> After testing out different scenarios, it appears to be only when the<br>
>> destination client is the <a href="http://webrtc.freeswitch.org/" target="_blank">webrtc.freeswitch.org</a> version of JsSIP there will<br>
>> be audio. Neither sipml5 demo or JsSIP demo website registered with<br>
>> <a href="http://webrtc.freeswitch.org/" target="_blank">webrtc.freeswitch.org</a> can generate audio. Something in the SDP maybe?<br>
>><br>
>> Thanks,<br>
>><br>
>> Henry<br>
>></div></div></blockquote></div></div></blockquote></div></div></div></div></div><br>_________________________________________________________________________<br>
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