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res_pjsip: Add "webrtc" configuration option

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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
This commit is contained in:
Kevin Harwell
2017-07-10 18:17:44 -05:00
parent 0f45c979a3
commit 7da6ddda30
9 changed files with 151 additions and 7 deletions
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63
res/res_pjsip_sdp_rtp.c
View File

@@ -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;