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| author | Kevin Harwell <kharwell@digium.com> | 2017-07-10 18:17:44 -0500 |
|---|---|---|
| committer | Kevin Harwell <kharwell@digium.com> | 2017-07-13 18:19:35 -0500 |
| commit | 7da6ddda30ab9291ec810fa88d4219145616bae8 (patch) | |
| tree | 89ad7fa5ae53b18a0a6412e85903ff7d8cd9d58b /configs | |
| parent | 0f45c979a3de00b320e05ba93309cf412e9e2702 (diff) | |
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
Diffstat (limited to 'configs')
| -rw-r--r-- | configs/samples/pjsip.conf.sample | 8 |
1 files changed, 8 insertions, 0 deletions
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample index c05938ea5..3c3e52a05 100644 --- a/configs/samples/pjsip.conf.sample +++ b/configs/samples/pjsip.conf.sample @@ -790,6 +790,14 @@ ; (default: 1) ;max_video_streams= ; The maximum number of allowed negotiated video streams ; (default: 1) +;webrtc= ; When set to "yes" this also enables the following values that are needed + ; for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport. + ; The following configuration settings also get defaulted as follows: + ; media_encryption=dtls + ; dtls_verify=fingerprint + ; dtls_setup=actpass + ; A dtls_cert_file and a dtls_ca_file still need to be specified. + ; Default for this option is "no" ;==========================AUTH SECTION OPTIONS========================= ;[auth] |