[asterisk-dev] icesupport in rtp.conf , but we only have chan_sip.

[Ian Wang]

Thank you Joshua, very beautiful, now webrtc is working very well.

Thanks,
Best Regards.


Ian WANG
Software Engineer
Fonality Pty Ltd(Australia)

office:  +6128484 2601 ext 3007
mobile: +61402524079

On Mon, Nov 7, 2016 at 10:20 PM, Joshua Colp <jcolp at digium.com> wrote:

> On Mon, Nov 7, 2016, at 02:14 AM, Ian Wang wrote:
> > hello,
> > I'm trying webrtc on Asterisk 14.1.1, I've found asterisk is missing ice
> > information in SDP which causing no audio issue on Chrome browser,
> >
> > in rtp.conf, I did configure the stun and turn server info, while in code
> > res_rtp_asterisk.c
> > it has following code , it seems I have to compile pjsip to have full ice
> > support,
> > but we have merged a lot of features in chan_sip.c, we don't want to use
> > pjsip now, is there any solution that we can continue using chan_sip only
> > but with full ice feature support , thanks.
>
> You don't need to switch to using chan_pjsip but you do need to install
> pjproject (of which PJSIP is a part). This is because pjproject also
> provides pjnath, which does ICE/STUN/TURN, that res_rtp_asterisk uses.
> The easiest way is to enable the bundle option to configure[1] which
> will automatically download and build it.
>
> [1]
> http://blogs.asterisk.org/2016/03/16/asterisk-13-8-0-
> now-easier-pjsip-install-method/
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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