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chan_sip: Add rtcp-mux support

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ASTERISK-26846 #close

Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639
pull/7/head
Sean Bright 8 years ago
parent 732367e806
commit 15aa3c0a23
4 changed files with 120 additions and 31 deletions
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7
UPGRADE.txt
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@ -27,9 +27,10 @@ From 14.3.0 to 14.4.0:
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

139
channels/chan_sip.c
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@ -1216,6 +1216,7 @@ 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_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);
@ -6011,7 +6012,7 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
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");
}
@ -6031,14 +6032,14 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
/* 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));
@ -7752,6 +7753,15 @@ static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_
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
@ -7780,7 +7790,7 @@ static int initialize_udptl(struct sip_pvt *p)
/* 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);
@ -8206,20 +8216,28 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
* 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) {
@ -10074,6 +10092,42 @@ static int has_media_stream(struct sip_pvt *p, enum media_type m)
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.
@ -10132,6 +10186,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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;
unsigned int numberofports;
@ -10662,6 +10720,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
} 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 */
@ -10683,6 +10743,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
} 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;
}
}
/* Text (T.140) specific scanning */
@ -10857,6 +10919,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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);
}
@ -10870,11 +10933,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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);
@ -10897,10 +10956,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* 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)
@ -10911,6 +10970,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* 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);
@ -10919,6 +10979,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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)
@ -11265,6 +11326,18 @@ static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_in
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;
}
return found;
}
static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
{
struct ast_rtp_engine_dtls *dtls;
@ -13632,6 +13705,12 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
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) {
@ -13999,18 +14078,18 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
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));
}
}
}
@ -21021,6 +21100,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
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 */
@ -21091,6 +21171,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
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)));
@ -21719,6 +21800,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
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) {
@ -30787,6 +30869,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} 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;
@ -33418,9 +33503,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
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));
@ -33432,9 +33517,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
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));
@ -33443,9 +33528,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
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));
}
}

3
channels/sip/include/sip.h
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@ -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

2
configs/samples/sip.conf.sample
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@ -1090,6 +1090,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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

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