[Asterisk-code-review] chan sip: Add rtcp-mux support (asterisk[14])
Joshua Colp
asteriskteam at digium.com
Sat Mar 18 05:38:03 CDT 2017
Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/5244 )
Change subject: chan_sip: Add rtcp-mux support
......................................................................
chan_sip: Add rtcp-mux support
ASTERISK-26846 #close
Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639
---
M UPGRADE.txt
M channels/chan_sip.c
M channels/sip/include/sip.h
M configs/samples/sip.conf.sample
4 files changed, 120 insertions(+), 31 deletions(-)
Approvals:
Richard Mudgett: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 2e72325..8682c72 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -26,9 +26,10 @@
res_rtp_asterisk:
- The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
- Data and Control Packets on a Single Port." So far, the only channel driver
- that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
- a PJSIP endpoint in pjsip.conf to enable the feature.
+ Data and Control Packets on a Single Port." For the PJSIP channel driver,
+ chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
+ to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
+ globally or on a per-peer basis in sip.conf.
New in 14.0.0
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2435d65..86c4bea 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1212,6 +1212,7 @@
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_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);
@@ -6018,7 +6019,7 @@
ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
ast_rtp_instance_set_qos(dialog->vrtp, global_tos_video, global_cos_video, "SIP VIDEO");
}
@@ -6038,14 +6039,14 @@
/* Do not timeout text as its not constant*/
ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
}
ast_rtp_instance_set_timeout(dialog->rtp, dialog->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->rtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
@@ -7759,6 +7760,15 @@
return res;
}
+enum sip_media_fds {
+ SIP_AUDIO_RTP_FD,
+ SIP_AUDIO_RTCP_FD,
+ SIP_VIDEO_RTP_FD,
+ SIP_VIDEO_RTCP_FD,
+ SIP_TEXT_RTP_FD,
+ SIP_UDPTL_FD,
+};
+
/*!
* \internal
* \brief Create and initialize UDPTL for the specified dialog
@@ -7787,7 +7797,7 @@
/* T38 can be supported by this dialog, create it and set the derived properties */
if ((p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, &bindaddr))) {
if (p->owner) {
- ast_channel_set_fd(p->owner, 5, ast_udptl_fd(p->udptl));
+ ast_channel_set_fd(p->owner, SIP_UDPTL_FD, ast_udptl_fd(p->udptl));
}
ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
@@ -8213,20 +8223,28 @@
* UDPTL is created as needed in the lifetime of a dialog, its file
* descriptor is set in initialize_udptl */
if (i->rtp) {
- ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
- ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTP_FD, ast_rtp_instance_fd(i->rtp, 0));
+ if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, -1);
+ } else {
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(i->rtp, 1));
+ }
ast_rtp_instance_set_write_format(i->rtp, fmt);
ast_rtp_instance_set_read_format(i->rtp, fmt);
}
if (needvideo && i->vrtp) {
- ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
- ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTP_FD, ast_rtp_instance_fd(i->vrtp, 0));
+ if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, -1);
+ } else {
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(i->vrtp, 1));
+ }
}
if (needtext && i->trtp) {
- ast_channel_set_fd(tmp, 4, ast_rtp_instance_fd(i->trtp, 0));
+ ast_channel_set_fd(tmp, SIP_TEXT_RTP_FD, ast_rtp_instance_fd(i->trtp, 0));
}
if (i->udptl) {
- ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
+ ast_channel_set_fd(tmp, SIP_UDPTL_FD, ast_udptl_fd(i->udptl));
}
if (state == AST_STATE_RING) {
@@ -10081,6 +10099,42 @@
return 0;
}
+static void configure_rtcp(struct sip_pvt *p, struct ast_rtp_instance *instance, int which, int remote_rtcp_mux)
+{
+ int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
+ int fd = -1;
+
+ if (local_rtcp_mux && remote_rtcp_mux) {
+ ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX);
+ } else {
+ ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
+ fd = ast_rtp_instance_fd(instance, 1);
+ }
+
+ if (p->owner) {
+ ast_channel_set_fd(p->owner, which, fd);
+ }
+}
+
+static void set_ice_components(struct sip_pvt *p, struct ast_rtp_instance *instance, int remote_rtcp_mux)
+{
+ struct ast_rtp_engine_ice *ice;
+ int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
+
+ ice = ast_rtp_instance_get_ice(instance);
+ if (!ice) {
+ return;
+ }
+
+ if (local_rtcp_mux && remote_rtcp_mux) {
+ /* We both support RTCP mux. Only one ICE component necessary */
+ ice->change_components(instance, 1);
+ } else {
+ /* They either don't support RTCP mux or we don't know if they do yet. */
+ ice->change_components(instance, 2);
+ }
+}
+
/*! \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.
@@ -10138,6 +10192,10 @@
/* SRTP */
int secure_audio = FALSE;
int secure_video = FALSE;
+
+ /* RTCP Multiplexing */
+ int remote_rtcp_mux_audio = FALSE;
+ int remote_rtcp_mux_video = FALSE;
/* Others */
int sendonly = -1;
@@ -10665,6 +10723,8 @@
processed = TRUE;
} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
processed = TRUE;
+ } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_audio)) {
+ processed = TRUE;
}
}
/* Video specific scanning */
@@ -10681,6 +10741,8 @@
processed_crypto = TRUE;
processed = TRUE;
} else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
+ processed = TRUE;
+ } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_video)) {
processed = TRUE;
}
}
@@ -10856,6 +10918,7 @@
if (sa && portno > 0) {
/* Start ICE negotiation here, only when it is response, and setting that we are conrolling agent,
as we are offerer */
+ set_ice_components(p, p->rtp, remote_rtcp_mux_audio);
if (req->method == SIP_RESPONSE) {
start_ice(p->rtp, 1);
}
@@ -10869,11 +10932,7 @@
ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
/* Ensure RTCP is enabled since it may be inactive
if we're coming back from a T.38 session */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
- /* Ensure audio RTCP reads are enabled */
- if (p->owner) {
- ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
- }
+ configure_rtcp(p, p->rtp, SIP_AUDIO_RTCP_FD, remote_rtcp_mux_audio);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -10896,10 +10955,10 @@
/* Prevent audio RTCP reads */
if (p->owner) {
- ast_channel_set_fd(p->owner, 1, -1);
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
}
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
} else {
ast_rtp_instance_stop(p->rtp);
if (debug)
@@ -10910,6 +10969,7 @@
/* Setup video address and port */
if (p->vrtp) {
if (vsa && vportno > 0) {
+ set_ice_components(p, p->vrtp, remote_rtcp_mux_video);
start_ice(p->vrtp, (req->method != SIP_RESPONSE) ? 0 : 1);
ast_sockaddr_set_port(vsa, vportno);
ast_rtp_instance_set_remote_address(p->vrtp, vsa);
@@ -10918,6 +10978,7 @@
ast_sockaddr_stringify(vsa));
}
ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
+ configure_rtcp(p, p->vrtp, SIP_VIDEO_RTCP_FD, remote_rtcp_mux_video);
} else {
ast_rtp_instance_stop(p->vrtp);
if (debug)
@@ -11258,6 +11319,18 @@
found = TRUE;
} else if (!strcasecmp(a, "ice-lite")) {
ice->ice_lite(instance);
+ found = TRUE;
+ }
+
+ return found;
+}
+
+static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested)
+{
+ int found = FALSE;
+
+ if (!strncasecmp(a, "rtcp-mux", 8)) {
+ *requested = TRUE;
found = TRUE;
}
@@ -13631,6 +13704,12 @@
add_dtls_to_sdp(p->rtp, &a_audio);
}
+
+ /* 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)) {
+ ast_str_append(&a_audio, 0, "a=rtcp-mux\r\n");
+ ast_str_append(&a_video, 0, "a=rtcp-mux\r\n");
+ }
}
if (add_t38) {
@@ -13998,18 +14077,18 @@
if (p->rtp) {
if (t38version) {
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
if (p->owner) {
/* Prevent audio RTCP reads */
- ast_channel_set_fd(p->owner, 1, -1);
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
}
} else if (ast_sockaddr_isnull(&p->redirip)) {
/* Enable RTCP since it will be inactive if we're coming back
* with this reinvite */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
if (p->owner) {
/* Enable audio RTCP reads */
- ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(p->rtp, 1));
}
}
}
@@ -21020,6 +21099,7 @@
ast_cli(fd, " Parkinglot : %s\n", peer->parkinglot);
ast_cli(fd, " Use Reason : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)));
ast_cli(fd, " Encryption : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)));
+ ast_cli(fd, " RTCP Mux : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX)));
ast_cli(fd, "\n");
peer = sip_unref_peer(peer, "sip_show_peer: sip_unref_peer: done with peer ptr");
} else if (peer && type == 1) { /* manager listing */
@@ -21090,6 +21170,7 @@
astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
astman_append(s, "SIP-Encryption: %s\r\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP) ? "Y" : "N");
+ astman_append(s, "SIP-RTCP-Mux: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX) ? "Y" : "N");
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -21718,6 +21799,7 @@
ast_cli(a->fd, " MOH Interpret: %s\n", default_mohinterpret);
ast_cli(a->fd, " MOH Suggest: %s\n", default_mohsuggest);
ast_cli(a->fd, " Voice Mail Extension: %s\n", default_vmexten);
+ ast_cli(a->fd, " RTCP Multiplexing: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[2], SIP_PAGE3_RTCP_MUX)));
if (realtimepeers || realtimeregs) {
@@ -30788,6 +30870,9 @@
} else if (!strcasecmp(v->name, "buggymwi")) {
ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
+ } else if (!strcasecmp(v->name, "rtcp_mux")) {
+ ast_set_flag(&mask[2], SIP_PAGE3_RTCP_MUX);
+ ast_set2_flag(&flags[2], ast_true(v->value), SIP_PAGE3_RTCP_MUX);
} else
res = 0;
@@ -33419,9 +33504,9 @@
if (p->rtp) {
/* Prevent audio RTCP reads */
- ast_channel_set_fd(chan, 1, -1);
+ ast_channel_set_fd(chan, SIP_AUDIO_RTCP_FD, -1);
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
}
} else if (!ast_sockaddr_isnull(&p->redirip)) {
memset(&p->redirip, 0, sizeof(p->redirip));
@@ -33433,9 +33518,9 @@
if (p->vrtp) {
/* Prevent video RTCP reads */
- ast_channel_set_fd(chan, 3, -1);
+ ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, -1);
/* Silence RTCP while video RTP is inactive */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
}
} else if (!ast_sockaddr_isnull(&p->vredirip)) {
memset(&p->vredirip, 0, sizeof(p->vredirip));
@@ -33444,9 +33529,9 @@
if (p->vrtp) {
/* Enable RTCP since it will be inactive if we're coming back
* from a reinvite */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
/* Enable video RTCP reads */
- ast_channel_set_fd(chan, 3, ast_rtp_instance_fd(p->vrtp, 1));
+ ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(p->vrtp, 1));
}
}
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index e511d13..86f8967 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -384,11 +384,12 @@
#define SIP_PAGE3_IGNORE_PREFCAPS (1 << 7) /*!< DP: Ignore prefcaps when setting up an outgoing call leg */
#define SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL (1 << 8) /*!< DGP: Stop telling the peer to start music on hold */
#define SIP_PAGE3_FORCE_AVP (1 << 9) /*!< DGP: Force 'RTP/AVP' for all streams, even DTLS */
+#define SIP_PAGE3_RTCP_MUX (1 << 10) /*!< DGP: Attempt to negotiate RFC 5761 RTCP multiplexing */
#define SIP_PAGE3_FLAGS_TO_COPY \
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
SIP_PAGE3_DIRECT_MEDIA_OUTGOING | SIP_PAGE3_USE_AVPF | SIP_PAGE3_ICE_SUPPORT | SIP_PAGE3_IGNORE_PREFCAPS | \
- SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP)
+ SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP | SIP_PAGE3_RTCP_MUX)
#define CHECK_AUTH_BUF_INITLEN 256
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
index 916e2d6..9b52ec0 100644
--- a/configs/samples/sip.conf.sample
+++ b/configs/samples/sip.conf.sample
@@ -1090,6 +1090,8 @@
; option may be specified at the global or peer scope.
;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
; media streams when appropriate, even if a DTLS stream is present.
+;rtcp_mux=yes ; Enable support for RFC 5761 RTCP multiplexing which is required for
+ ; WebRTC support
; ---------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
--
To view, visit https://gerrit.asterisk.org/5244
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Sean Bright <sean.bright at gmail.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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