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

Joshua Colp jcolp at digium.com
Mon Nov 7 05:20:34 CST 2016


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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