Wow this has been asked and answered about 27 times already... Yes Video is supported, no video transcoding is not supported... Now as far as video freezing/doing other strange things, this is going to depend on what the other end is doing along with various other network issues...
Example: if you are calling into mod_conference to do a multi-party video conference with VP8 its not going to work all that well as VP8 doesn’t have keyframes that are sent on a regular basis as with other codecs like H264/H263 and there is currently limited to no support in mod_conference to automattically handle swapping from participant A to participant B, mixed with the fact that all video on FreeSWITCH is passthru only, there is no “brandy bunch” screens etc...
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?
Thanks,
Henry
On Fri, Jul 5, 2013 at 3:15 PM, Michael Jerris <mike@jerris.com> wrote:
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.
Mike
On Jul 5, 2013, at 6:05 PM, Henry Huang <red.rain.seven@gmail.com> wrote:
I confirm that the changes made by Iwan worked and gave me audio. Thanks, Iwan.
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?
Thanks,
Henry
On Fri, Jul 5, 2013 at 2:27 PM, Iwan Budi Kusnanto <ibk@labhijau.net> wrote:
>> register to webrtc.freeswitch.org <http://webrtc.freeswitch.org/> then I will be able to replicate the noHenry,
I can make it jssip demo works by modify this line
this.peerConnection = new
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);
into
constraints["optional"] = [];
constraints["optional"].push({'DtlsSrtpKeyAgreement': 'true'});
this.peerConnection = new
JsSIP.WebRTC.RTCPeerConnection({'iceServers': servers}, constraints);
On Sat, Jul 6, 2013 at 1:12 AM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:
> Maybe because we hacked in dtls support?
>
>
>
> On Fri, Jul 5, 2013 at 12:58 PM, Henry Huang <red.rain.seven@gmail.com>
> wrote:
>>
>> I think I have found the issue. If I use the JsSIP website demo to
>> audio issue on webrtc.freeswitch.org <http://webrtc.freeswitch.org/>
>>
>> After testing out different scenarios, it appears to be only when the
>> destination client is the webrtc.freeswitch.org <http://webrtc.freeswitch.org/> version of JsSIP there will
>> be audio. Neither sipml5 demo or JsSIP demo website registered with
>> webrtc.freeswitch.org <http://webrtc.freeswitch.org/> can generate audio. Something in the SDP maybe?
>>
>> Thanks,
>>
>> Henry
>>
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_________________________________________________________________________
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