[asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients

[Matthew Jordan]

On Thu, Oct 2, 2014 at 10:18 AM, Olli Heiskanen
<ohjelmistoarkkitehti at gmail.com> wrote:
> Hi,
>
> Thanks Eric for your reply, yes I know Asterisk replaces the sdp, however it
> should create ice lines when calling to a webrtc client, which it is
> currently not doing.
>
> To recap my problem (check previous messages for details); I have 2 webrtc
> clients (sip.js on chrome) with realtime information that appears to be
> correct. When calling from A to B, INVITE coming to Asterisk contains
> correct sdp, but when the INVITE leaves Asterisk, the sdp lacks ice lines.
>

Unfortunately, I can't reproduce this. We've been running a lot of
tests with a variety of SIP clients over the past week here at SIPit -
both with and without ICE - and I haven't had a single instance of
Asterisk failing to provide any ICE candidates when it is properly
configured to do so.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org