[Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

[Ken Rice]

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...



On 7/8/13 11:24 AM, "Henry Huang" <red.rain.seven at gmail.com> wrote:

> 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 at 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 at 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 at labhijau.net> wrote:
>>>> Henry,
>>>> 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 at gmail.com> wrote:
>>>>> > Maybe because we hacked in dtls support?
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Fri, Jul 5, 2013 at 12:58 PM, Henry Huang <red.rain.seven at gmail.com>
>>>>> > wrote:
>>>>>> >>
>>>>>> >> I think I have found the issue. If I use the JsSIP website demo to
>>>>>> >> register to webrtc.freeswitch.org <http://webrtc.freeswitch.org/>
>>>>>> then I will be able to replicate the no
>>>>>> >> 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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> 
> 
> 
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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