[Spce-user] WebRTC to Legacy SIP no 2 way audio/video

[Daniel Grotti]

Jon,
you may need to tune and change:

use_rtpproxy: Always with rtpproxy as additional/only ICE candidate
transport_protocol: RTP/SAVPF (encrypted SRTP with RTCP feedback)


Try to play with those two parameters, cause it depends on the browser
you are using.

--
Daniel Grotti
VoIP Engineer


Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com

On 08/19/2015 02:41 PM, Jon Bonilla (Manwe) wrote:
> El Wed, 19 Aug 2015 11:54:10 +0200
> Daniel Grotti <dgrotti at sipwise.com> escribió:
> 
>> Hi Jon,
>> have you had a looks at this :
>> https://www.linkedin.com/pulse/how-enable-webrtc-sipprovider-daniel-grotti?_mSplash=1
>>
>>
>>
>> Maybe it may help.
>> P.S. some configuration may have changed due to the new version of
>> Chrome/Firefox.
>>
>>
> 
> Hi Daniel
> 
> Yes, I changed the preferences for webrtc subscribers. 
>