[Asterisk-code-review] chan sip: Better ICE handling for RTCP-MUX (asterisk[master])
Sean Bright
asteriskteam at digium.com
Fri May 19 10:11:15 CDT 2017
Sean Bright has uploaded a new change for review. ( https://gerrit.asterisk.org/5652 )
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 or emit 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, 50 insertions(+), 14 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/52/5652/1
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 930dc0f..8ea3120 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1215,14 +1215,14 @@
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);
static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec);
static int process_sdp_a_image(const char *a, struct sip_pvt *p);
-static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
+static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf, int rtcp_mux);
static void add_dtls_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
static void start_ice(struct ast_rtp_instance *instance, int offer);
static void add_codec_to_sdp(const struct sip_pvt *p, struct ast_format *codec,
@@ -10128,6 +10128,24 @@
}
}
+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.
@@ -10270,13 +10288,13 @@
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;
}
@@ -10316,6 +10334,7 @@
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;
@@ -10334,6 +10353,9 @@
}
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) {
@@ -10701,7 +10723,7 @@
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;
@@ -10726,7 +10748,7 @@
}
/* 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;
@@ -10749,7 +10771,7 @@
}
/* 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;
@@ -11271,7 +11293,7 @@
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;
@@ -11291,6 +11313,12 @@
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;
@@ -12960,7 +12988,7 @@
}
/*! \brief Add ICE attributes to SDP */
-static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf)
+static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf, int rtcp_mux)
{
struct ast_rtp_engine_ice *ice = ast_rtp_instance_get_ice(instance);
const char *username, *password;
@@ -12983,6 +13011,12 @@
i = ao2_iterator_init(candidates, 0);
while ((candidate = ao2_iterator_next(&i))) {
+ /* Don't emit RTCP candidates if we are offering RTCP-MUX */
+ if (rtcp_mux && candidate->id > 1) {
+ ao2_ref(candidate, -1);
+ continue;
+ }
+
ast_str_append(a_buf, 0, "a=candidate:%s %u %s %d ", candidate->foundation, candidate->id, candidate->transport, candidate->priority);
ast_str_append(a_buf, 0, "%s ", ast_sockaddr_stringify_host(&candidate->address));
@@ -13420,6 +13454,8 @@
int min_video_packet_size = 0;
int min_text_packet_size = 0;
+ int rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
+
struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
/* Set the SDP session name */
@@ -13545,7 +13581,7 @@
if (!doing_directmedia) {
if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
- add_ice_to_sdp(p->vrtp, &a_video);
+ add_ice_to_sdp(p->vrtp, &a_video, rtcp_mux);
}
add_dtls_to_sdp(p->vrtp, &a_video);
@@ -13569,7 +13605,7 @@
if (!doing_directmedia) {
if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
- add_ice_to_sdp(p->trtp, &a_text);
+ add_ice_to_sdp(p->trtp, &a_text, rtcp_mux);
}
add_dtls_to_sdp(p->trtp, &a_text);
@@ -13695,7 +13731,7 @@
if (!doing_directmedia) {
if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
- add_ice_to_sdp(p->rtp, &a_audio);
+ add_ice_to_sdp(p->rtp, &a_audio, rtcp_mux);
/* Start ICE negotiation, and setting that we are controlled agent,
as this is response to offer */
if (resp->method == SIP_RESPONSE) {
@@ -13707,7 +13743,7 @@
}
/* If we've got rtcp-mux enabled, just unconditionally offer it in all SDPs */
- if (ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ if (rtcp_mux) {
ast_str_append(&a_audio, 0, "a=rtcp-mux\r\n");
ast_str_append(&a_video, 0, "a=rtcp-mux\r\n");
}
--
To view, visit https://gerrit.asterisk.org/5652
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Sean Bright <sean.bright at gmail.com>
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