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

[Olli Heiskanen]

Hi,

Thanks Matthew for trying to reproduce the problem, I appreciate your
efforts very much.

There must be something off in my setup in one way or another. I could just
discard this server and build a new one, but I think it's not good practice
to leave a problem unsolved, so I'll continue trying to figure this out.
One thing I noticed - don't know if it's relevant or not - due to a repo
mismatch, I had problems with updating libgdiplus and libgdiplus-devel
package, had to disable a repo and reinstall those and my mono installation
(which is making me lose my hair).

Is there a way to debug Asterisk itself? Or find the code that parses the
outbound sdp? I figured there must be an if statement or more that
determines whether or not to parse the ice lines into the sdp body. Finding
that/those statements that produce the kind of sdp I'm seeing Asterisk send
out, might tell something about what's wrong with my setup. As my c is not
exactly fluent I wasn't sure which code files to search, can you guys help
out with that?

cheers,
Olli



2014-10-03 11:31 GMT+03:00 Matthew Jordan <mjordan at digium.com>:

> 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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141007/1cf4d76d/attachment.html>