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:
https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF
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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