diff options
author | Jenkins2 <jenkins2@gerrit.asterisk.org> | 2017-07-17 15:16:30 -0500 |
---|---|---|
committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2017-07-17 15:16:30 -0500 |
commit | 647f539e1569ad27c46352e58be9447e21c23923 (patch) | |
tree | a4d42c9bce7c5087b2bbc709af9815322cf599a9 /res | |
parent | 29af7d5558d325e262fc033fd1399e79bcefb0e0 (diff) | |
parent | 7da6ddda30ab9291ec810fa88d4219145616bae8 (diff) |
Merge "res_pjsip: Add "webrtc" configuration option"
Diffstat (limited to 'res')
-rw-r--r-- | res/res_pjsip.c | 24 | ||||
-rw-r--r-- | res/res_pjsip.exports.in | 1 | ||||
-rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 27 | ||||
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 63 | ||||
-rw-r--r-- | res/res_pjsip_session.c | 9 |
5 files changed, 118 insertions, 6 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index ee5c5fe5e..02112113c 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -1010,6 +1010,18 @@ underlying transport. Note that enabling bundle will also enable the rtcp_mux option. </para></description> </configOption> + <configOption name="webrtc" default="no"> + <synopsis>Defaults and enables some options that are relevant to WebRTC</synopsis> + <description><para> + When set to "yes" this also enables the following values that are needed in + order for basic WebRTC support to work: rtcp_mux, use_avpf, ice_support, and + use_received_transport. The following configuration settings also get defaulted + as follows:</para> + <para>media_encryption=dtls</para> + <para>dtls_verify=fingerprint</para> + <para>dtls_setup=actpass</para> + </description> + </configOption> </configObject> <configObject name="auth"> <synopsis>Authentication type</synopsis> @@ -4244,6 +4256,18 @@ void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size) dest[chars_to_copy] = '\0'; } +int ast_copy_pj_str2(char **dest, const pj_str_t *src) +{ + int res = ast_asprintf(dest, "%.*s", (int)pj_strlen(src), pj_strbuf(src)); + + if (res < 0) { + *dest = NULL; + } + + return res; +} + + int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype) { pjsip_media_type compare; diff --git a/res/res_pjsip.exports.in b/res/res_pjsip.exports.in index 8b62abbfe..4adecd419 100644 --- a/res/res_pjsip.exports.in +++ b/res/res_pjsip.exports.in @@ -2,6 +2,7 @@ global: LINKER_SYMBOL_PREFIXast_sip_*; LINKER_SYMBOL_PREFIXast_copy_pj_str; + LINKER_SYMBOL_PREFIXast_copy_pj_str2; LINKER_SYMBOL_PREFIXast_pjsip_rdata_get_endpoint; local: *; diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index c60173721..9f9de36fa 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1363,8 +1363,30 @@ static int sip_endpoint_apply_handler(const struct ast_sorcery *sorcery, void *o return -1; } - if (endpoint->media.bundle) { - endpoint->media.rtcp_mux = 1; + endpoint->media.rtcp_mux |= endpoint->media.bundle; + + /* + * If webrtc has been enabled then enable those attributes, and default + * some, that are needed in order for webrtc to work. + */ + endpoint->media.bundle |= endpoint->media.webrtc; + endpoint->media.rtcp_mux |= endpoint->media.webrtc; + endpoint->media.rtp.use_avpf |= endpoint->media.webrtc; + endpoint->media.rtp.ice_support |= endpoint->media.webrtc; + endpoint->media.rtp.use_received_transport |= endpoint->media.webrtc; + + if (endpoint->media.webrtc) { + endpoint->media.rtp.encryption = AST_SIP_MEDIA_ENCRYPT_DTLS; + endpoint->media.rtp.dtls_cfg.enabled = 1; + endpoint->media.rtp.dtls_cfg.default_setup = AST_RTP_DTLS_SETUP_ACTPASS; + endpoint->media.rtp.dtls_cfg.verify = AST_RTP_DTLS_VERIFY_FINGERPRINT; + + if (ast_strlen_zero(endpoint->media.rtp.dtls_cfg.certfile) || + (ast_strlen_zero(endpoint->media.rtp.dtls_cfg.cafile))) { + ast_log(LOG_ERROR, "WebRTC can't be enabled on endpoint '%s' - a DTLS cert " + "or ca file has not been specified", ast_sorcery_object_get_id(endpoint)); + return -1; + } } return 0; @@ -1990,6 +2012,7 @@ int ast_res_pjsip_initialize_configuration(void) ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_audio_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_audio_streams)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_video_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_video_streams)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "bundle", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.bundle)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "webrtc", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.webrtc)); if (ast_sip_initialize_sorcery_transport()) { ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n"); diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 4ec811528..a2e7f8f92 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -1025,6 +1025,65 @@ static void process_ssrc_attributes(struct ast_sip_session *session, struct ast_ } } +static void process_msid_attribute(struct ast_sip_session *session, + struct ast_sip_session_media *session_media, pjmedia_sdp_media *media) +{ + pjmedia_sdp_attr *attr; + + if (!session->endpoint->media.webrtc) { + return; + } + + attr = pjmedia_sdp_media_find_attr2(media, "msid", NULL); + if (attr) { + ast_free(session_media->msid); + ast_copy_pj_str2(&session_media->msid, &attr->value); + } +} + +static void add_msid_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; + + if (!session->endpoint->media.webrtc) { + return; + } + + if (ast_strlen_zero(session_media->msid)) { + char uuid1[AST_UUID_STR_LEN], uuid2[AST_UUID_STR_LEN]; + + if (ast_asprintf(&session_media->msid, "{%s} {%s}", + ast_uuid_generate_str(uuid1, sizeof(uuid1)), + ast_uuid_generate_str(uuid2, sizeof(uuid2))) < 0) { + session_media->msid = NULL; + return; + } + } + + attr = pjmedia_sdp_attr_create(pool, "msid", pj_cstr(&stmp, session_media->msid)); + pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr); +} + +static void add_rtcp_fb_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; + + if (!session->endpoint->media.webrtc || session_media->type != AST_MEDIA_TYPE_VIDEO) { + return; + } + + /* + * For now just automatically add it the stream even though it hasn't + * necessarily been negotiated. + */ + attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* ccm fir")); + pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr); +} + /*! \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, @@ -1068,7 +1127,7 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, } process_ssrc_attributes(session, session_media, stream); - + process_msid_attribute(session, session_media, stream); session_media_transport = ast_sip_session_media_get_transport(session, session_media); if (session_media_transport == session_media || !session_media->bundled) { @@ -1527,6 +1586,8 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as } add_ssrc_to_stream(session, session_media, pool, media); + add_msid_to_stream(session, session_media, pool, media); + add_rtcp_fb_to_stream(session, session_media, pool, media); /* Add the media stream to the SDP */ sdp->media[sdp->media_count++] = media; diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c index 315db6df5..fe3680f3b 100644 --- a/res/res_pjsip_session.c +++ b/res/res_pjsip_session.c @@ -395,6 +395,7 @@ static void session_media_dtor(void *obj) } ast_free(session_media->mid); + ast_free(session_media->msid); } struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_session *session, @@ -3573,15 +3574,17 @@ static int add_bundle_groups(struct ast_sip_session *session, pj_pool_t *pool, p int index, mid_id; struct sip_session_media_bundle_group *bundle_group; + if (session->endpoint->media.webrtc) { + attr = pjmedia_sdp_attr_create(pool, "msid-semantic", pj_cstr(&stmp, "WMS *")); + pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr); + } + if (!session->endpoint->media.bundle) { return 0; } memset(bundle_groups, 0, sizeof(bundle_groups)); - attr = pjmedia_sdp_attr_create(pool, "msid-semantic", pj_cstr(&stmp, "WMS *")); - pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr); - /* Build the bundle group layout so we can then add it to the SDP */ for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) { struct ast_sip_session_media *session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index); |