diff options
| author | Joshua Colp <jcolp@digium.com> | 2016-10-23 12:38:59 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2016-10-26 12:48:57 +0000 |
| commit | aed6c219a3b21d212cea28425e0144eb52a3db70 (patch) | |
| tree | b0cc140b5f46ca0617b8b3a875c8f0171888b361 /res/res_pjsip_sdp_rtp.c | |
| parent | 7c79b057dd3b0a06000b7bdd0583648bfc643b96 (diff) | |
pjsip: Fix a few media bugs with reinvites and asymmetric payloads.
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
Diffstat (limited to 'res/res_pjsip_sdp_rtp.c')
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index a69aa1a74..13a71d478 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -385,6 +385,11 @@ static int set_caps(struct ast_sip_session *session, session->dsp = NULL; } } + + if (ast_channel_is_bridged(session->channel)) { + ast_channel_set_unbridged_nolock(session->channel, 1); + } + ast_channel_unlock(session->channel); } |