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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
