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.
*#ifdef HAVE_PJPROJECT*
/* Create an ICE session for ICE negotiation */
if (icesupport) {
ast_debug(3, "Creating ICE session %s (%d) for RTP instance '%p'\n",
ast_sockaddr_stringify(addr), x, instance);
if (ice_create(instance, addr, x, 0)) {
ast_log(LOG_NOTICE, "Failed to start ICE session\n");
} else {
rtp->ice_port = x;
ast_sockaddr_copy(&rtp->ice_original_rtp_addr, addr);
}
}
#endif
Thanks,
Best Regards.
Ian WANG
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