[asterisk-commits] chan sip: Better ICE handling for RTCP-MUX (asterisk[14])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue May 23 08:49:41 CDT 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5651 )
Change subject: chan_sip: Better ICE handling for RTCP-MUX
......................................................................
chan_sip: Better ICE handling for RTCP-MUX
If we are offered or are offering RTCP-MUX, don't consider RTCP ICE
candidates. This confuses certain browsers (current Firefox for
example) and causes intial audio setup delays.
ASTERISK-26982 #close
Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91
---
M channels/chan_sip.c
1 file changed, 36 insertions(+), 8 deletions(-)
Approvals:
George Joseph: Looks good to me, approved; Verified
Jenkins2: Approved for Submit
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index ff2e5ba..5affee6 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1211,7 +1211,7 @@
static int process_sdp_o(const char *o, struct sip_pvt *p);
static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
static int process_sdp_a_sendonly(const char *a, int *sendonly);
-static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
+static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance, int rtcp_mux);
static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested);
static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec);
@@ -10135,6 +10135,24 @@
}
}
+static int has_media_level_attribute(int start, struct sip_request *req, const char *attr)
+{
+ int next = start;
+ char type;
+ const char *value;
+
+ /* We don't care about the return result here */
+ get_sdp_iterate(&next, req, "m");
+
+ while ((type = get_sdp_line(&start, next, req, &value)) != '\0') {
+ if (type == 'a' && !strcasecmp(value, attr)) {
+ return 1;
+ }
+ }
+
+ return 0;
+}
+
/*! \brief Process SIP SDP offer, select formats and activate media channels
If offer is rejected, we will not change any properties of the call
Return 0 on success, a negative value on errors.
@@ -10277,13 +10295,13 @@
else if (process_sdp_a_image(value, p))
processed = TRUE;
- if (process_sdp_a_ice(value, p, p->rtp)) {
+ if (process_sdp_a_ice(value, p, p->rtp, 0)) {
processed = TRUE;
}
- if (process_sdp_a_ice(value, p, p->vrtp)) {
+ if (process_sdp_a_ice(value, p, p->vrtp, 0)) {
processed = TRUE;
}
- if (process_sdp_a_ice(value, p, p->trtp)) {
+ if (process_sdp_a_ice(value, p, p->trtp, 0)) {
processed = TRUE;
}
@@ -10323,6 +10341,7 @@
int image = FALSE;
int text = FALSE;
int processed_crypto = FALSE;
+ int rtcp_mux_offered = 0;
char protocol[18] = {0,};
unsigned int x;
struct ast_rtp_engine_dtls *dtls;
@@ -10341,6 +10360,9 @@
}
AST_LIST_INSERT_TAIL(&p->offered_media, offer, next);
offer->type = SDP_UNKNOWN;
+
+ /* We need to check for this ahead of time */
+ rtcp_mux_offered = has_media_level_attribute(iterator, req, "rtcp-mux");
/* Check for 'audio' media offer */
if (strncmp(m, "audio ", 6) == 0) {
@@ -10708,7 +10730,7 @@
case 'a':
/* Audio specific scanning */
if (audio) {
- if (process_sdp_a_ice(value, p, p->rtp)) {
+ if (process_sdp_a_ice(value, p, p->rtp, rtcp_mux_offered)) {
processed = TRUE;
} else if (process_sdp_a_dtls(value, p, p->rtp)) {
processed_crypto = TRUE;
@@ -10729,7 +10751,7 @@
}
/* Video specific scanning */
else if (video) {
- if (process_sdp_a_ice(value, p, p->vrtp)) {
+ if (process_sdp_a_ice(value, p, p->vrtp, rtcp_mux_offered)) {
processed = TRUE;
} else if (process_sdp_a_dtls(value, p, p->vrtp)) {
processed_crypto = TRUE;
@@ -10748,7 +10770,7 @@
}
/* Text (T.140) specific scanning */
else if (text) {
- if (process_sdp_a_ice(value, p, p->trtp)) {
+ if (process_sdp_a_ice(value, p, p->trtp, rtcp_mux_offered)) {
processed = TRUE;
} else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
processed = TRUE;
@@ -11270,7 +11292,7 @@
return found;
}
-static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
+static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance, int rtcp_mux_offered)
{
struct ast_rtp_engine_ice *ice;
int found = FALSE;
@@ -11290,6 +11312,12 @@
found = TRUE;
} else if (sscanf(a, "candidate: %31s %30u %3s %30u %23s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport, (unsigned *)&candidate.priority,
address, &port, cand_type, relay_address, &relay_port) >= 7) {
+
+ if (rtcp_mux_offered && ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX) && candidate.id > 1) {
+ /* If we support RTCP-MUX and they offered it, don't consider RTCP candidates */
+ return TRUE;
+ }
+
candidate.foundation = foundation;
candidate.transport = transport;
--
To view, visit https://gerrit.asterisk.org/5651
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Sean Bright <sean.bright at gmail.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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