diff options
author | Jenkins2 <jenkins2@gerrit.asterisk.org> | 2017-07-13 14:40:11 -0500 |
---|---|---|
committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2017-07-13 14:40:11 -0500 |
commit | 0f45c979a3de00b320e05ba93309cf412e9e2702 (patch) | |
tree | 1852402245ee52adb65acc5d47b1ab13857aaea0 /res/res_pjsip_sdp_rtp.c | |
parent | e83b9d141a416ab8c0b1fcfcd29d73abf2ca04c9 (diff) | |
parent | 065c3005ad920f5fe2cedcf062e38b8e28eeb015 (diff) |
Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."
Diffstat (limited to 'res/res_pjsip_sdp_rtp.c')
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 411 |
1 files changed, 294 insertions, 117 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index a49130868..4ec811528 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -317,6 +317,7 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp static int set_caps(struct ast_sip_session *session, struct ast_sip_session_media *session_media, + struct ast_sip_session_media *session_media_transport, const struct pjmedia_sdp_media *stream, int is_offer, struct ast_stream *asterisk_stream) { @@ -376,6 +377,24 @@ static int set_caps(struct ast_sip_session *session, ast_stream_set_formats(asterisk_stream, joint); + /* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */ + if (session_media_transport != session_media && session_media->bundled) { + int index; + + for (index = 0; index < ast_format_cap_count(joint); ++index) { + struct ast_format *format = ast_format_cap_get_format(joint, index); + int rtp_code; + + /* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for + * things as the format is guaranteed to have a payload already. + */ + rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0); + ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format); + + ao2_ref(format, -1); + } + } + if (session->channel && ast_sip_session_is_pending_stream_default(session, asterisk_stream)) { ast_channel_lock(session->channel); ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN); @@ -496,7 +515,8 @@ static pjmedia_sdp_attr* generate_fmtp_attr(pj_pool_t *pool, struct ast_format * } /*! \brief Function which adds ICE attributes to a media stream */ -static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media) +static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media, + unsigned int include_candidates) { struct ast_rtp_engine_ice *ice; struct ao2_container *candidates; @@ -506,8 +526,7 @@ static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_se struct ao2_iterator it_candidates; struct ast_rtp_engine_ice_candidate *candidate; - if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp)) || - !(candidates = ice->get_local_candidates(session_media->rtp))) { + if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) { return; } @@ -521,6 +540,15 @@ static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_se media->attr[media->attr_count++] = attr; } + if (!include_candidates) { + return; + } + + candidates = ice->get_local_candidates(session_media->rtp); + if (!candidates) { + return; + } + it_candidates = ao2_iterator_init(candidates, 0); for (; (candidate = ao2_iterator_next(&it_candidates)); ao2_ref(candidate, -1)) { struct ast_str *attr_candidate = ast_str_create(128); @@ -940,6 +968,63 @@ static void set_ice_components(struct ast_sip_session *session, struct ast_sip_s } } +/*! \brief Function which adds ssrc attributes to a media stream */ +static void add_ssrc_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media) +{ + pj_str_t stmp; + pjmedia_sdp_attr *attr; + char tmp[128]; + + if (!session->endpoint->media.bundle || session_media->bundle_group == -1) { + return; + } + + snprintf(tmp, sizeof(tmp), "%u cname:%s", ast_rtp_instance_get_ssrc(session_media->rtp), ast_rtp_instance_get_cname(session_media->rtp)); + attr = pjmedia_sdp_attr_create(pool, "ssrc", pj_cstr(&stmp, tmp)); + media->attr[media->attr_count++] = attr; +} + +/*! \brief Function which processes ssrc attributes in a stream */ +static void process_ssrc_attributes(struct ast_sip_session *session, struct ast_sip_session_media *session_media, + const struct pjmedia_sdp_media *remote_stream) +{ + int index; + + if (!session->endpoint->media.bundle) { + return; + } + + for (index = 0; index < remote_stream->attr_count; ++index) { + pjmedia_sdp_attr *attr = remote_stream->attr[index]; + char attr_value[pj_strlen(&attr->value) + 1]; + char *ssrc_attribute_name, *ssrc_attribute_value = NULL; + unsigned int ssrc; + + /* We only care about ssrc attributes */ + if (pj_strcmp2(&attr->name, "ssrc")) { + continue; + } + + ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value)); + + if ((ssrc_attribute_name = strchr(attr_value, ' '))) { + /* This has an actual attribute */ + *ssrc_attribute_name++ = '\0'; + ssrc_attribute_value = strchr(ssrc_attribute_name, ':'); + if (ssrc_attribute_value) { + /* Values are actually optional according to the spec */ + *ssrc_attribute_value++ = '\0'; + } + } + + if (sscanf(attr_value, "%30u", &ssrc) < 1) { + continue; + } + + ast_rtp_instance_set_remote_ssrc(session_media->rtp, ssrc); + } +} + /*! \brief Function which negotiates an incoming media stream */ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const pjmedia_sdp_session *sdp, @@ -948,6 +1033,7 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, char host[NI_MAXHOST]; RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free); pjmedia_sdp_media *stream = sdp->media[index]; + struct ast_sip_session_media *session_media_transport; enum ast_media_type media_type = session_media->type; enum ast_sip_session_media_encryption encryption = AST_SIP_MEDIA_ENCRYPT_NONE; int res; @@ -981,38 +1067,51 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, return -1; } - session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(stream, "rtcp-mux", NULL) != NULL); - set_ice_components(session, session_media); + process_ssrc_attributes(session, session_media, stream); - enable_rtcp(session, session_media, stream); + session_media_transport = ast_sip_session_media_get_transport(session, session_media); - res = setup_media_encryption(session, session_media, sdp, stream); - if (res) { - if (!session->endpoint->media.rtp.encryption_optimistic || - !pj_strncmp2(&stream->desc.transport, "RTP/SAVP", 8)) { - /* If optimistic encryption is disabled and crypto should have been enabled - * but was not this session must fail. This must also fail if crypto was - * required in the offer but could not be set up. - */ - return -1; + if (session_media_transport == session_media || !session_media->bundled) { + /* If this media session is carrying actual traffic then set up those aspects */ + session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(stream, "rtcp-mux", NULL) != NULL); + set_ice_components(session, session_media); + + enable_rtcp(session, session_media, stream); + + res = setup_media_encryption(session, session_media, sdp, stream); + if (res) { + if (!session->endpoint->media.rtp.encryption_optimistic || + !pj_strncmp2(&stream->desc.transport, "RTP/SAVP", 8)) { + /* If optimistic encryption is disabled and crypto should have been enabled + * but was not this session must fail. This must also fail if crypto was + * required in the offer but could not be set up. + */ + return -1; + } + /* There is no encryption, sad. */ + session_media->encryption = AST_SIP_MEDIA_ENCRYPT_NONE; } - /* There is no encryption, sad. */ - session_media->encryption = AST_SIP_MEDIA_ENCRYPT_NONE; - } - /* If we've been explicitly configured to use the received transport OR if - * encryption is on and crypto is present use the received transport. - * This is done in case of optimistic because it may come in as RTP/AVP or RTP/SAVP depending - * on the configuration of the remote endpoint (optimistic themselves or mandatory). - */ - if ((session->endpoint->media.rtp.use_received_transport) || - ((encryption == AST_SIP_MEDIA_ENCRYPT_SDES) && !res)) { - pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport); - } + /* If we've been explicitly configured to use the received transport OR if + * encryption is on and crypto is present use the received transport. + * This is done in case of optimistic because it may come in as RTP/AVP or RTP/SAVP depending + * on the configuration of the remote endpoint (optimistic themselves or mandatory). + */ + if ((session->endpoint->media.rtp.use_received_transport) || + ((encryption == AST_SIP_MEDIA_ENCRYPT_SDES) && !res)) { + pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport); + } + } else { + /* This is bundled with another session, so mark it as such */ + ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp); - if (set_caps(session, session_media, stream, 1, asterisk_stream)) { + enable_rtcp(session, session_media, stream); + } + + if (set_caps(session, session_media, session_media_transport, stream, 1, asterisk_stream)) { return 0; } + return 1; } @@ -1032,6 +1131,7 @@ static int add_crypto_to_stream(struct ast_sip_session *session, static const pj_str_t STR_PASSIVE = { "passive", 7 }; static const pj_str_t STR_ACTPASS = { "actpass", 7 }; static const pj_str_t STR_HOLDCONN = { "holdconn", 8 }; + enum ast_rtp_dtls_setup setup; switch (session_media->encryption) { case AST_SIP_MEDIA_ENCRYPT_NONE: @@ -1085,7 +1185,16 @@ static int add_crypto_to_stream(struct ast_sip_session *session, break; } - switch (dtls->get_setup(session_media->rtp)) { + /* If this is an answer we need to use our current state, if it's an offer we need to use + * the configured value. + */ + if (pjmedia_sdp_neg_get_state(session->inv_session->neg) != PJMEDIA_SDP_NEG_STATE_DONE) { + setup = dtls->get_setup(session_media->rtp); + } else { + setup = session->endpoint->media.rtp.dtls_cfg.default_setup; + } + + switch (setup) { case AST_RTP_DTLS_SETUP_ACTIVE: attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTIVE); media->attr[media->attr_count++] = attr; @@ -1100,7 +1209,6 @@ static int add_crypto_to_stream(struct ast_sip_session *session, break; case AST_RTP_DTLS_SETUP_HOLDCONN: attr = pjmedia_sdp_attr_create(pool, "setup", &STR_HOLDCONN); - media->attr[media->attr_count++] = attr; break; default: break; @@ -1152,6 +1260,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as int rtp_code; RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup); enum ast_media_type media_type = session_media->type; + struct ast_sip_session_media *session_media_transport; int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) && ast_format_cap_count(session->direct_media_cap); @@ -1195,68 +1304,106 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as return -1; } - set_ice_components(session, session_media); - enable_rtcp(session, session_media, NULL); + /* If this stream has not been bundled already it is new and we need to ensure there is no SSRC conflict */ + if (session_media->bundle_group != -1 && !session_media->bundled) { + for (index = 0; index < sdp->media_count; ++index) { + struct ast_sip_session_media *other_session_media; - /* Crypto has to be added before setting the media transport so that SRTP is properly - * set up according to the configuration. This ends up changing the media transport. - */ - if (add_crypto_to_stream(session, session_media, pool, media)) { - return -1; - } + other_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index); + if (!other_session_media->rtp || other_session_media->bundle_group != session_media->bundle_group) { + continue; + } - if (pj_strlen(&session_media->transport)) { - /* If a transport has already been specified use it */ - media->desc.transport = session_media->transport; - } else { - media->desc.transport = pj_str(ast_sdp_get_rtp_profile( - /* Optimistic encryption places crypto in the normal RTP/AVP profile */ - !session->endpoint->media.rtp.encryption_optimistic && - (session_media->encryption == AST_SIP_MEDIA_ENCRYPT_SDES), - session_media->rtp, session->endpoint->media.rtp.use_avpf, - session->endpoint->media.rtp.force_avp)); + if (ast_rtp_instance_get_ssrc(session_media->rtp) == ast_rtp_instance_get_ssrc(other_session_media->rtp)) { + ast_rtp_instance_change_source(session_media->rtp); + /* Start the conflict check over again */ + index = -1; + continue; + } + } } - media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn)); - if (!media->conn) { - return -1; - } + session_media_transport = ast_sip_session_media_get_transport(session, session_media); - /* Add connection level details */ - if (direct_media_enabled) { - hostip = ast_sockaddr_stringify_fmt(&session_media->direct_media_addr, AST_SOCKADDR_STR_ADDR); - } else if (ast_strlen_zero(session->endpoint->media.address)) { - hostip = ast_sip_get_host_ip_string(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET()); - } else { - hostip = session->endpoint->media.address; - } + if (session_media_transport == session_media || !session_media->bundled) { + set_ice_components(session, session_media); + enable_rtcp(session, session_media, NULL); - if (ast_strlen_zero(hostip)) { - ast_log(LOG_ERROR, "No local host IP available for stream %s\n", - ast_codec_media_type2str(session_media->type)); - return -1; - } + /* Crypto has to be added before setting the media transport so that SRTP is properly + * set up according to the configuration. This ends up changing the media transport. + */ + if (add_crypto_to_stream(session, session_media, pool, media)) { + return -1; + } - media->conn->net_type = STR_IN; - /* Assume that the connection will use IPv4 until proven otherwise */ - media->conn->addr_type = STR_IP4; - pj_strdup2(pool, &media->conn->addr, hostip); + if (pj_strlen(&session_media->transport)) { + /* If a transport has already been specified use it */ + media->desc.transport = session_media->transport; + } else { + media->desc.transport = pj_str(ast_sdp_get_rtp_profile( + /* Optimistic encryption places crypto in the normal RTP/AVP profile */ + !session->endpoint->media.rtp.encryption_optimistic && + (session_media->encryption == AST_SIP_MEDIA_ENCRYPT_SDES), + session_media->rtp, session->endpoint->media.rtp.use_avpf, + session->endpoint->media.rtp.force_avp)); + } - if (!ast_strlen_zero(session->endpoint->media.address)) { - pj_sockaddr ip; + media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn)); + if (!media->conn) { + return -1; + } - if ((pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &media->conn->addr, &ip) == PJ_SUCCESS) && - (ip.addr.sa_family == pj_AF_INET6())) { - media->conn->addr_type = STR_IP6; + /* Add connection level details */ + if (direct_media_enabled) { + hostip = ast_sockaddr_stringify_fmt(&session_media->direct_media_addr, AST_SOCKADDR_STR_ADDR); + } else if (ast_strlen_zero(session->endpoint->media.address)) { + hostip = ast_sip_get_host_ip_string(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET()); + } else { + hostip = session->endpoint->media.address; } - } - /* Add ICE attributes and candidates */ - add_ice_to_stream(session, session_media, pool, media); + if (ast_strlen_zero(hostip)) { + ast_log(LOG_ERROR, "No local host IP available for stream %s\n", + ast_codec_media_type2str(session_media->type)); + return -1; + } + + media->conn->net_type = STR_IN; + /* Assume that the connection will use IPv4 until proven otherwise */ + media->conn->addr_type = STR_IP4; + pj_strdup2(pool, &media->conn->addr, hostip); + + if (!ast_strlen_zero(session->endpoint->media.address)) { + pj_sockaddr ip; + + if ((pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &media->conn->addr, &ip) == PJ_SUCCESS) && + (ip.addr.sa_family == pj_AF_INET6())) { + media->conn->addr_type = STR_IP6; + } + } + + /* Add ICE attributes and candidates */ + add_ice_to_stream(session, session_media, pool, media, 1); + + ast_rtp_instance_get_local_address(session_media->rtp, &addr); + media->desc.port = direct_media_enabled ? ast_sockaddr_port(&session_media->direct_media_addr) : (pj_uint16_t) ast_sockaddr_port(&addr); + media->desc.port_count = 1; + } else { + pjmedia_sdp_media *bundle_group_stream = sdp->media[session_media_transport->stream_num]; + + /* As this is in a bundle group it shares the same details as the group instance */ + media->desc.transport = bundle_group_stream->desc.transport; + media->conn = bundle_group_stream->conn; + media->desc.port = bundle_group_stream->desc.port; + + if (add_crypto_to_stream(session, session_media_transport, pool, media)) { + return -1; + } - ast_rtp_instance_get_local_address(session_media->rtp, &addr); - media->desc.port = direct_media_enabled ? ast_sockaddr_port(&session_media->direct_media_addr) : (pj_uint16_t) ast_sockaddr_port(&addr); - media->desc.port_count = 1; + add_ice_to_stream(session, session_media_transport, pool, media, 0); + + enable_rtcp(session, session_media, NULL); + } if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", @@ -1278,10 +1425,23 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as continue; } - if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0)) == -1) { - ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format)); - ao2_ref(format, -1); - continue; + /* If this stream is not a transport we need to use the transport codecs structure for payload management to prevent + * conflicts. + */ + if (session_media_transport != session_media) { + if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media_transport->rtp), 1, format, 0)) == -1) { + ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format)); + ao2_ref(format, -1); + continue; + } + /* Our instance has to match the payload number though */ + ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media->rtp), rtp_code, format); + } else { + if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0)) == -1) { + ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format)); + ao2_ref(format, -1); + continue; + } } if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) { @@ -1332,6 +1492,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as } } + /* If no formats were actually added to the media stream don't add it to the SDP */ if (!media->desc.fmt_count) { return 1; @@ -1365,6 +1526,8 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr); } + add_ssrc_to_stream(session, session_media, pool, media); + /* Add the media stream to the SDP */ sdp->media[sdp->media_count++] = media; @@ -1425,6 +1588,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, enum ast_media_type media_type = session_media->type; char host[NI_MAXHOST]; int res; + struct ast_sip_session_media *session_media_transport; if (!session->channel) { return 1; @@ -1441,48 +1605,60 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, return -1; } - session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(remote_stream, "rtcp-mux", NULL) != NULL); - set_ice_components(session, session_media); + process_ssrc_attributes(session, session_media, remote_stream); - enable_rtcp(session, session_media, remote_stream); + session_media_transport = ast_sip_session_media_get_transport(session, session_media); - res = setup_media_encryption(session, session_media, remote, remote_stream); - if (!session->endpoint->media.rtp.encryption_optimistic && res) { - /* If optimistic encryption is disabled and crypto should have been enabled but was not - * this session must fail. - */ - return -1; - } + if (session_media_transport == session_media || !session_media->bundled) { + session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(remote_stream, "rtcp-mux", NULL) != NULL); + set_ice_components(session, session_media); - if (!remote_stream->conn && !remote->conn) { - return 1; - } + enable_rtcp(session, session_media, remote_stream); - ast_copy_pj_str(host, remote_stream->conn ? &remote_stream->conn->addr : &remote->conn->addr, sizeof(host)); + res = setup_media_encryption(session, session_media, remote, remote_stream); + if (!session->endpoint->media.rtp.encryption_optimistic && res) { + /* If optimistic encryption is disabled and crypto should have been enabled but was not + * this session must fail. + */ + return -1; + } - /* Ensure that the address provided is valid */ - if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) { - /* The provided host was actually invalid so we error out this negotiation */ - return -1; - } + if (!remote_stream->conn && !remote->conn) { + return 1; + } - /* Apply connection information to the RTP instance */ - ast_sockaddr_set_port(addrs, remote_stream->desc.port); - ast_rtp_instance_set_remote_address(session_media->rtp, addrs); - if (set_caps(session, session_media, remote_stream, 0, asterisk_stream)) { - return 1; - } + ast_copy_pj_str(host, remote_stream->conn ? &remote_stream->conn->addr : &remote->conn->addr, sizeof(host)); - ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback); - ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 0), - media_session_rtp_read_callback); - if (!session->endpoint->media.rtcp_mux || !session_media->remote_rtcp_mux) { - ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 1), - media_session_rtcp_read_callback); + /* Ensure that the address provided is valid */ + if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) { + /* The provided host was actually invalid so we error out this negotiation */ + return -1; + } + + /* Apply connection information to the RTP instance */ + ast_sockaddr_set_port(addrs, remote_stream->desc.port); + ast_rtp_instance_set_remote_address(session_media->rtp, addrs); + + ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback); + ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 0), + media_session_rtp_read_callback); + if (!session->endpoint->media.rtcp_mux || !session_media->remote_rtcp_mux) { + ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 1), + media_session_rtcp_read_callback); + } + + /* If ICE support is enabled find all the needed attributes */ + process_ice_attributes(session, session_media, remote, remote_stream); + } else { + /* This is bundled with another session, so mark it as such */ + ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp); + ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback); + enable_rtcp(session, session_media, remote_stream); } - /* If ICE support is enabled find all the needed attributes */ - process_ice_attributes(session, session_media, remote, remote_stream); + if (set_caps(session, session_media, session_media_transport, remote_stream, 0, asterisk_stream)) { + return 1; + } /* Set the channel uniqueid on the RTP instance now that it is becoming active */ ast_channel_lock(session->channel); @@ -1490,6 +1666,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, ast_channel_unlock(session->channel); /* Ensure the RTP instance is active */ + ast_rtp_instance_set_stream_num(session_media->rtp, ast_stream_get_position(asterisk_stream)); ast_rtp_instance_activate(session_media->rtp); /* audio stream handles music on hold */ |