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 <[email protected]> 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 > > -- > _____________________________________________________________________ > -- 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 >
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