res_pjsip: Add "webrtc" configuration option
authorKevin Harwell <kharwell@digium.com>
Mon, 10 Jul 2017 23:17:44 +0000 (18:17 -0500)
committerKevin Harwell <kharwell@digium.com>
Thu, 13 Jul 2017 23:19:35 +0000 (18:19 -0500)
commit7da6ddda30ab9291ec810fa88d4219145616bae8
tree89ad7fa5ae53b18a0a6412e85903ff7d8cd9d58b
parent0f45c979a3de00b320e05ba93309cf412e9e2702
res_pjsip: Add "webrtc" configuration option

This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:

  rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
  media_encryption=dtls
  dtls_verify=fingerprint
  dtls_setup=actpass

When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.

Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.

ASTERISK-27119 #close

Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
channels/chan_pjsip.c
configs/samples/pjsip.conf.sample
include/asterisk/res_pjsip.h
include/asterisk/res_pjsip_session.h
res/res_pjsip.c
res/res_pjsip.exports.in
res/res_pjsip/pjsip_configuration.c
res/res_pjsip_sdp_rtp.c
res/res_pjsip_session.c