[asterisk-users] Confbridge SFU for Asterisk 15

Carlos Chavez cursor at telecomab.mx
Wed Nov 15 11:30:52 CST 2017

On 11/15/17 11:10 AM, Joshua Colp wrote:

> On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote:
>> On 11/14/17 5:23 PM, Joshua Colp wrote:
>>> On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote:
>>>> Trace with 3 clients.  We can hear each other but no video.
>>>> https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz
>>> Do you see anything in the Javascript console of the browser? We are
>>> adding the needed media streams by sending a reinvite to the
>>> participants but we don't get any response, which means for some reason
>>> the browser may not have liked what we provided.
>> This is what I get on the console:
>> new session - outgoing - [object Object]
>> cyber_mega_phone.js:78:3
>> ontrack: audio - 8b7fca5e-bb67-4e8c-8bdb-84fb80ac4cc0 stream
>> 66e4250b-c196-4482-a347-d12772ef865d
>> cyber_mega_phone.js:111:4
>> Streams: added 66e4250b-c196-4482-a347-d12772ef865d
>> cyber_mega_phone.js:225:3
>> ontrack: video - ad836e20-c0c9-423f-9c42-0aef19c5ca32 stream
>> 66e4250b-c196-4482-a347-d12772ef865d
>> cyber_mega_phone.js:111:4
>> confirmed: adding local stream {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
>> cyber_mega_phone.js:84:5
>> Streams: added {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
>> cyber_mega_phone.js:225:3
>> RTCPeerConnection.getLocalStreams/getRemoteStreams are deprecated. Use
>> RTCPeerConnection.getSenders/getReceivers instead.
>> cyber_mega_phone.js:82:17
>> ICE failed, add a STUN server and see about:webrtc for more details
> Looks like for some reason it failed to successfully do ICE negotiation
> potentially on the newly added remote streams. Why that is is
> environment specific - but the problem does seem to be on the web
> browser/client side, not in Asterisk itself. You'd need to figure out
> why.
> This is one of the annoyances of WebRTC - the browser can be a black box
> at time and when things go wrong (like this) it's hard to dig and figure
> out what is up.
Here is more information from the browser about the session:

On Asterisk I have icesupport=true in rtp.conf and ice_support=yes on the endpoint.  I have configured a STUN server in both rtp.conf and res_stun_monitor.conf

Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161

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