diff options
author | Matthew Jordan <mjordan@digium.com> | 2013-08-23 15:42:27 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2013-08-23 15:42:27 +0000 |
commit | 4d348e853cbd9ba7bc976487bfcb352a84e5ece0 (patch) | |
tree | fdf289e34cd706884aed7a262409fc3cdcba9bd1 /res/res_pjsip_sdp_rtp.c | |
parent | e31bd332b83f0245ce8bd6626279e1b9c683ec18 (diff) |
Add pass through support for Opus and VP8; Opus format attribute negotiation
This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip_sdp_rtp.c')
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 14 |
1 files changed, 13 insertions, 1 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index c97c0cb40..be5d59f06 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -849,7 +849,9 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as char tmp[512]; pj_str_t stmp; pjmedia_sdp_attr *attr; - int index = 0, min_packet_size = 0, noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0; + int index = 0; + int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0; + int min_packet_size = 0, max_packet_size = 0; int rtp_code; struct ast_format format; RAII_VAR(struct ast_format_cap *, caps, NULL, ast_format_cap_destroy); @@ -951,6 +953,10 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as if (fmt.cur_ms && ((fmt.cur_ms < min_packet_size) || !min_packet_size)) { min_packet_size = fmt.cur_ms; } + + if (fmt.max_ms && ((fmt.max_ms < max_packet_size) || !max_packet_size)) { + max_packet_size = fmt.max_ms; + } } } @@ -983,6 +989,12 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as media->attr[media->attr_count++] = attr; } + if (max_packet_size) { + snprintf(tmp, sizeof(tmp), "%d", max_packet_size); + attr = pjmedia_sdp_attr_create(pool, "maxptime", pj_cstr(&stmp, tmp)); + media->attr[media->attr_count++] = attr; + } + /* Add the sendrecv attribute - we purposely don't keep track because pjmedia-sdp will automatically change our offer for us */ attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr); attr->name = STR_SENDRECV; |