diff options
| author | Joshua Colp <jcolp@digium.com> | 2017-06-30 18:55:57 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2017-07-13 14:47:50 +0000 |
| commit | 065c3005ad920f5fe2cedcf062e38b8e28eeb015 (patch) | |
| tree | 6f10effde768ae19faa7342b1eeef1004cf2fddc /channels | |
| parent | e1c0e14fac7cdca276a0fd61fa891ea54f4ab5d8 (diff) | |
res_rtp_asterisk / res_pjsip: Add support for BUNDLE.
BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.
This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.
For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.
ASTERISK-27118
Change-Id: I96c0920b9f9aca7382256484765a239017973c11
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_pjsip.c | 2 |
1 files changed, 0 insertions, 2 deletions
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c index 0e4468cdc..3de980a52 100644 --- a/channels/chan_pjsip.c +++ b/channels/chan_pjsip.c @@ -792,8 +792,6 @@ static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast) return f; } - f->stream_num = callback_state->session->stream_num; - if (f->frametype != AST_FRAME_VOICE || callback_state->session != session->active_media_state->default_session[callback_state->session->type]) { return f; |