diff options
| author | Mark Michelson <mmichelson@digium.com> | 2013-07-18 19:25:51 +0000 |
|---|---|---|
| committer | Mark Michelson <mmichelson@digium.com> | 2013-07-18 19:25:51 +0000 |
| commit | c47787feab34d9572b41ebe7148c169b52362fbd (patch) | |
| tree | 85410d3780938b31d9605f6d05cef9a744c3932d /res/res_sip_sdp_rtp.c | |
| parent | 3c86832f9f271c8d479cf956155424fda512c76b (diff) | |
Add a bunch of options from sip.conf to res_sip.conf
For a complete list of the options added, see the review linked
at the bottom of this commit message.
(closes issue ASTERISK-21506)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2671
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_sip_sdp_rtp.c')
| -rw-r--r-- | res/res_sip_sdp_rtp.c | 13 |
1 files changed, 12 insertions, 1 deletions
diff --git a/res/res_sip_sdp_rtp.c b/res/res_sip_sdp_rtp.c index bc150ed4a..b299ed758 100644 --- a/res/res_sip_sdp_rtp.c +++ b/res/res_sip_sdp_rtp.c @@ -108,7 +108,8 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me { struct ast_rtp_engine_ice *ice; - if (!(session_media->rtp = ast_rtp_instance_new("asterisk", sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) { + if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->rtp_engine, sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) { + ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->rtp_engine); return -1; } @@ -132,6 +133,16 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND); } + if (!strcmp(session_media->stream_type, STR_AUDIO) && + (session->endpoint->tos_audio || session->endpoint->cos_video)) { + ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->tos_audio, + session->endpoint->cos_audio, "SIP RTP Audio"); + } else if (!strcmp(session_media->stream_type, STR_VIDEO) && + (session->endpoint->tos_video || session->endpoint->cos_video)) { + ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->tos_video, + session->endpoint->cos_video, "SIP RTP Video"); + } + return 0; } |