Merge "res_pjsip: Add "webrtc" configuration option"

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 */

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]

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
  *

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;
 };
 
 /*!

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;

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:
 		*;

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 |= endpoint->media.bundle;
-		endpoint->media.rtcp_mux = 1;
+
+	/*
+	 * 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");

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;

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);