[asterisk-commits] res pjsip: Add "webrtc" configuration option (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jul 17 15:16:30 CDT 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5988 )
Change subject: res_pjsip: Add "webrtc" configuration option
......................................................................
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
---
M channels/chan_pjsip.c
M configs/samples/pjsip.conf.sample
M include/asterisk/res_pjsip.h
M include/asterisk/res_pjsip_session.h
M res/res_pjsip.c
M res/res_pjsip.exports.in
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
9 files changed, 151 insertions(+), 7 deletions(-)
Approvals:
Benjamin Keith Ford: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, but someone else must approve
Matthew Fredrickson: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 931b608..f009943 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1595,7 +1595,9 @@
/* 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 c05938e..3c3e52a 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 d499d55..cf366cb 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -690,6 +690,8 @@
unsigned int max_video_streams;
/*! Use BUNDLE */
unsigned int bundle;
+ /*! Enable webrtc settings and defaults */
+ unsigned int webrtc;
};
/*!
@@ -2061,6 +2063,24 @@
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
*
* The function may ONLY be called on out-of-dialog requests or responses. For
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index eae29de..eae11af 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -105,6 +105,8 @@
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 ee5c5fe..0211211 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 @@
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 8b62abb..4adecd4 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 c601737..9f9de36 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1363,8 +1363,30 @@
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 @@
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 4ec8115..a2e7f8f 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -1025,6 +1025,65 @@
}
}
+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 @@
}
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 @@
}
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 315db6d..fe3680f 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -395,6 +395,7 @@
}
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,
@@ -3573,14 +3574,16 @@
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) {
--
To view, visit https://gerrit.asterisk.org/5988
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
Gerrit-Change-Number: 5988
Gerrit-PatchSet: 5
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-commits/attachments/20170717/76e09e5f/attachment-0001.html>
More information about the asterisk-commits
mailing list