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authorMatthew Jordan <mjordan@digium.com>2013-08-23 15:42:27 +0000
committerMatthew Jordan <mjordan@digium.com>2013-08-23 15:42:27 +0000
commit4d348e853cbd9ba7bc976487bfcb352a84e5ece0 (patch)
treefdf289e34cd706884aed7a262409fc3cdcba9bd1 /res/res_pjsip_sdp_rtp.c
parente31bd332b83f0245ce8bd6626279e1b9c683ec18 (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.c14
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;