chan_sip: Add rtcp-mux support
authorSean Bright <sean.bright@gmail.com>
Wed, 8 Mar 2017 01:28:18 +0000 (20:28 -0500)
committerSean Bright <sean.bright@gmail.com>
Fri, 17 Mar 2017 13:36:06 +0000 (07:36 -0600)
ASTERISK-26846 #close

Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639

UPGRADE.txt
channels/chan_sip.c
channels/sip/include/sip.h
configs/samples/sip.conf.sample

index 2275580..1afacf2 100644 (file)
@@ -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
 
index d158b0d..f659a44 100644 (file)
@@ -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));
                }
        }
 
index e511d13..86f8967 100644 (file)
 #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
 
index 916e2d6..9b52ec0 100644 (file)
@@ -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