From 7da6ddda30ab9291ec810fa88d4219145616bae8 Mon Sep 17 00:00:00 2001 From: Kevin Harwell Date: Mon, 10 Jul 2017 18:17:44 -0500 Subject: [PATCH] res_pjsip: Add "webrtc" configuration option This patch creates a new configuration option called "webrtc". When enabled it defaults and enables the following options that are needed in order for webrtc to work in Asterisk: rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled media_encryption=dtls dtls_verify=fingerprint dtls_setup=actpass When "webrtc" is enabled, this patch also parses the "msid" media level attribute from an SDP. It will also appropriately add it onto the outgoing session when applicable. Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. ASTERISK-27119 #close Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd --- channels/chan_pjsip.c | 4 +- configs/samples/pjsip.conf.sample | 8 ++++ include/asterisk/res_pjsip.h | 20 +++++++++ include/asterisk/res_pjsip_session.h | 2 + res/res_pjsip.c | 24 +++++++++++ res/res_pjsip.exports.in | 1 + res/res_pjsip/pjsip_configuration.c | 27 +++++++++++- res/res_pjsip_sdp_rtp.c | 63 +++++++++++++++++++++++++++- res/res_pjsip_session.c | 9 ++-- 9 files changed, 151 insertions(+), 7 deletions(-) diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c index 931b608390..f009943ed9 100644 --- a/channels/chan_pjsip.c +++ b/channels/chan_pjsip.c @@ -1595,7 +1595,9 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi /* FIXME: Only use this for VP8. Additional work would have to be done to * fully support other video codecs */ - if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL) { + if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL || + (channel->session->endpoint->media.webrtc && + ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h264) != AST_FORMAT_CMP_NOT_EQUAL)) { /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the * RTP engine would provide a way to externally write/schedule RTCP * packets */ diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample index c05938ea5f..3c3e52a053 100644 --- a/configs/samples/pjsip.conf.sample +++ b/configs/samples/pjsip.conf.sample @@ -790,6 +790,14 @@ ; (default: 1) ;max_video_streams= ; The maximum number of allowed negotiated video streams ; (default: 1) +;webrtc= ; When set to "yes" this also enables the following values that are needed + ; for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport. + ; The following configuration settings also get defaulted as follows: + ; media_encryption=dtls + ; dtls_verify=fingerprint + ; dtls_setup=actpass + ; A dtls_cert_file and a dtls_ca_file still need to be specified. + ; Default for this option is "no" ;==========================AUTH SECTION OPTIONS========================= ;[auth] diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h index d499d5514b..cf366cbab3 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -690,6 +690,8 @@ struct ast_sip_endpoint_media_configuration { unsigned int max_video_streams; /*! Use BUNDLE */ unsigned int bundle; + /*! Enable webrtc settings and defaults */ + unsigned int webrtc; }; /*! @@ -2060,6 +2062,24 @@ int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text); */ void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size); +/*! + * \brief Create and copy a pj_str_t into a standard character buffer. + * + * pj_str_t is not NULL-terminated. Any place that expects a NULL- + * terminated string needs to have the pj_str_t copied into a separate + * buffer. + * + * Copies the pj_str_t contents into a newly allocated buffer pointed to + * by dest. NULL-terminates the buffer. + * + * \note Caller is responsible for freeing the allocated memory. + * + * \param dest [out] The destination buffer + * \param src The pj_str_t to copy + * \retval Number of characters copied or negative value on error + */ +int ast_copy_pj_str2(char **dest, const pj_str_t *src); + /*! * \brief Get the looked-up endpoint on an out-of dialog request or response * diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h index eae29de046..eae11af434 100644 --- a/include/asterisk/res_pjsip_session.h +++ b/include/asterisk/res_pjsip_session.h @@ -105,6 +105,8 @@ struct ast_sip_session_media { int bundle_group; /*! \brief Whether this stream is currently bundled or not */ unsigned int bundled; + /*! \brief RTP/Media streams association identifier */ + char *msid; }; /*! diff --git a/res/res_pjsip.c b/res/res_pjsip.c index ee5c5fe5e9..02112113cd 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. + + Defaults and enables some options that are relevant to WebRTC + + 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: + media_encryption=dtls + dtls_verify=fingerprint + dtls_setup=actpass + + Authentication type @@ -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 8b62abbfe4..4adecd419c 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 c60173721b..9f9de36faa 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 4ec8115288..a2e7f8f922 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 315db6df5d..fe3680f3b7 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);