diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 931b608390584ebbba4b0d0935712ec6e31e32d5..f009943ed9a719ad12088b3f43213cb6186ee13d 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 c05938ea5fdcc00ae26a029831b17a002ed169e2..3c3e52a053f73335ae8ab34c1feece5c6f4b7918 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 d499d5514bfca5fa97e835adb1573724b809ea4b..cf366cbab36ba7ed2743f9c9aedd1c179358583e 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 eae29de046753d9825377d48636be83e9ed93911..eae11af43488dd43692f6c742d6d0899e78b0203 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 ee5c5fe5e90dbeff63782e19eeb5db874b09cf16..02112113cd7d8ca31a6eab2abf80e8eab4082bbf 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 8b62abbfe45b84b19d89f93f71ac332a76eecf08..4adecd419c225ead7709d147fd25f3eb70186e3e 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 c60173721b94e29729bd86ebee93d9f3278961fc..9f9de36faa3dfb854ab6aa1a32a641b4d554eea7 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 4ec811528807ad0958a9ebe4ba84ea95a75d5cf6..a2e7f8f922326076e9fcec88b304da049ccb01d5 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 315db6df5db1c30ded0b08fdd61fbe022c772b66..fe3680f3b785eaf0f159afc0ee13c8032953f35d 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);