Skip to content

Commit

Permalink
feat(sctp): add sctp object, and SRS_SCTP macro
Browse files Browse the repository at this point in the history
  • Loading branch information
@johzzy
johzzy committed Apr 3, 2023
1 parent 7176142 commit e511fd9
Show file tree
Hide file tree
Showing 13 changed files with 1,077 additions and 9 deletions.
29 changes: 29 additions & 0 deletions trunk/configure
View file
Original file line number Diff line number Diff line change
Expand Up @@ -154,6 +154,11 @@ if [[ $SRS_RTC == YES ]]; then
LibSrtpRoot="${SRS_OBJS}/srtp2/include"; LibSrtpFile="${SRS_OBJS}/srtp2/lib/libsrtp2.a"
fi

# sctp
if [[ $SRS_SCTP == YES ]]; then
LibSctpRoot="${SRS_OBJS}/sctp/include"; LibSctpFile="${SRS_OBJS}/sctp/lib/libusrsctp.a"
fi

# FFMPEG for WebRTC transcoding, such as aac to opus.
if [[ $SRS_FFMPEG_FIT == YES ]]; then
LibFfmpegRoot="${SRS_OBJS}/ffmpeg/include"
Expand Down Expand Up @@ -289,6 +294,9 @@ fi
if [[ $SRS_RTC == YES ]]; then
ModuleLibIncs+=(${LibSrtpRoot})
fi
if [[ $SRS_SCTP == YES ]]; then
ModuleLibIncs+=(${LibSctpRoot})
fi
if [[ $SRS_FFMPEG_FIT == YES ]]; then
ModuleLibIncs+=("${LibFfmpegRoot[*]}")
fi
Expand Down Expand Up @@ -319,6 +327,9 @@ fi
if [[ $SRS_GB28181 == YES ]]; then
MODULE_FILES+=("srs_app_gb28181")
fi
if [[ $SRS_SCTP == YES ]]; then
MODULE_FILES+=("srs_app_sctp")
fi

DEFINES=""
# add each modules for app
Expand All @@ -340,6 +351,9 @@ fi
if [[ $SRS_FFMPEG_FIT == YES ]]; then
ModuleLibIncs+=("${LibFfmpegRoot[*]}")
fi
if [[ $SRS_SCTP == YES ]]; then
ModuleLibIncs+=(${LibSctpRoot})
fi
MODULE_FILES=("srs_main_server")
SERVER_INCS="src/main"; MODULE_DIR=${SERVER_INCS} . $SRS_WORKDIR/auto/modules.sh
SERVER_OBJS="${MODULE_OBJS[@]}"
Expand All @@ -354,6 +368,9 @@ fi
if [[ $SRS_FFMPEG_FIT == YES ]]; then
ModuleLibIncs+=("${LibFfmpegRoot[*]}")
fi
if [[ $SRS_SCTP == YES ]]; then
ModuleLibIncs+=(${LibSctpRoot})
fi
MODULE_FILES=()
DEFINES=""
# add each modules for main
Expand Down Expand Up @@ -384,6 +401,9 @@ fi
if [[ $SRS_FFMPEG_FIT == YES ]]; then
ModuleLibFiles+=("${LibFfmpegFile[*]}")
fi
if [[ $SRS_SCTP == YES ]]; then
ModuleLibFiles+=(${LibSctpFile})
fi
if [[ $SRS_SRT == YES ]]; then
ModuleLibFiles+=("${LibSRTfile[*]}")
fi
Expand All @@ -396,6 +416,9 @@ fi
if [[ $SRS_FFMPEG_FIT == YES ]]; then
ModuleLibIncs+=("${LibFfmpegRoot[*]}")
fi
if [[ $SRS_SCTP == YES ]]; then
ModuleLibIncs+=(${LibSctpRoot})
fi
if [[ $SRS_SRT == YES ]]; then
ModuleLibIncs+=(${LibSRTRoot})
MODULE_OBJS="${MODULE_OBJS} ${SRT_OBJS[@]}"
Expand Down Expand Up @@ -449,6 +472,9 @@ if [[ $SRS_UTEST == YES ]]; then
if [[ $SRS_FFMPEG_FIT == YES ]]; then
ModuleLibIncs+=("${LibFfmpegRoot[*]}")
fi
if [[ $SRS_SCTP == YES ]]; then
ModuleLibIncs+=(${LibSctpRoot})
fi
if [[ $SRS_SRT == YES ]]; then
ModuleLibIncs+=("${LibSRTRoot[*]}")
fi
Expand All @@ -459,6 +485,9 @@ if [[ $SRS_UTEST == YES ]]; then
if [[ $SRS_FFMPEG_FIT == YES ]]; then
ModuleLibFiles+=("${LibFfmpegFile[*]}")
fi
if [[ $SRS_SCTP == YES ]]; then
ModuleLibFiles+=(${LibSctpFile})
fi
if [[ $SRS_SRT == YES ]]; then
ModuleLibFiles+=("${LibSRTfile[*]}")
fi
Expand Down
260 changes: 257 additions & 3 deletions trunk/src/app/srs_app_rtc_api.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -263,12 +263,12 @@ srs_error_t SrsGoApiRtcPlay::check_remote_sdp(const SrsSdp& remote_sdp)
}

for (std::vector<SrsMediaDesc>::const_iterator iter = remote_sdp.media_descs_.begin(); iter != remote_sdp.media_descs_.end(); ++iter) {
if (iter->type_ != "audio" && iter->type_ != "video") {
if (iter->type_ != "audio" && iter->type_ != "video" && iter->type_ != "application") {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "unsupport media type=%s", iter->type_.c_str());
}

if (! iter->rtcp_mux_) {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "now only suppor rtcp-mux");
if (!iter->rtcp_mux_ && iter->type_ != "application") {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "now media only support rtcp-mux");
}

if (iter->sendonly_) {
Expand Down Expand Up @@ -312,6 +312,260 @@ srs_error_t SrsGoApiRtcPlay::http_hooks_on_play(SrsRequest* req)
return err;
}

#ifdef SRS_SCTP

SrsGoApiRtcDataChannel::SrsGoApiRtcDataChannel(SrsRtcServer* server)
{
server_ = server;
}

SrsGoApiRtcDataChannel::~SrsGoApiRtcDataChannel()
{
}


// Request:
// POST /rtc/v1/data/
// {
// "sdp":"offer...", "streamurl":"webrtc://r.ossrs.net/live/livestream",
// "api":'http...", "clientip":"..."
// }
// Response:
// {"sdp":"answer...", "sid":"..."}
// @see https://github.com/rtcdn/rtcdn-draft
srs_error_t SrsGoApiRtcDataChannel::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
{
srs_error_t err = srs_success;

SrsJsonObject* res = SrsJsonAny::object();
SrsAutoFree(SrsJsonObject, res);

if ((err = do_serve_http(w, r, res)) != srs_success) {
srs_warn("RTC error %s", srs_error_desc(err).c_str()); srs_freep(err);
return srs_api_response_code(w, r, SRS_CONSTS_HTTP_BadRequest);
}

return srs_api_response(w, r, res->dumps());
}

srs_error_t SrsGoApiRtcDataChannel::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res)
{
srs_error_t err = srs_success;

// For each RTC session, we use short-term HTTP connection.
SrsHttpHeader* hdr = w->header();
hdr->set("Connection", "Close");

// Parse req, the request json object, from body.
SrsJsonObject* req = NULL;
SrsAutoFree(SrsJsonObject, req);
if (true) {
string req_json;
if ((err = r->body_read_all(req_json)) != srs_success) {
return srs_error_wrap(err, "read body");
}

SrsJsonAny* json = SrsJsonAny::loads(req_json);
if (!json || !json->is_object()) {
return srs_error_new(ERROR_RTC_API_BODY, "invalid body %s", req_json.c_str());
}

req = json->to_object();
}

// Fetch params from req object.
SrsJsonAny* prop = NULL;
if ((prop = req->ensure_property_string("sdp")) == NULL) {
return srs_error_wrap(err, "not sdp");
}
string remote_sdp_str = prop->to_str();

if ((prop = req->ensure_property_string("streamurl")) == NULL) {
return srs_error_wrap(err, "not streamurl");
}
string streamurl = prop->to_str();

string clientip;
if ((prop = req->ensure_property_string("clientip")) != NULL) {
clientip = prop->to_str();
}

string api;
if ((prop = req->ensure_property_string("api")) != NULL) {
api = prop->to_str();
}

// The RTC user config object.
SrsRtcUserConfig ruc;
ruc.req_->ip = clientip;

srs_parse_rtmp_url(streamurl, ruc.req_->tcUrl, ruc.req_->stream);

srs_discovery_tc_url(ruc.req_->tcUrl, ruc.req_->schema, ruc.req_->host, ruc.req_->vhost,
ruc.req_->app, ruc.req_->stream, ruc.req_->port, ruc.req_->param);

// discovery vhost, resolve the vhost from config
SrsConfDirective* parsed_vhost = _srs_config->get_vhost(ruc.req_->vhost);
if (parsed_vhost) {
ruc.req_->vhost = parsed_vhost->arg0();
}

// For client to specifies the EIP of server.
string eip = r->query_get("eip");
// For client to specifies whether encrypt by SRTP.
string srtp = r->query_get("encrypt");
string dtls = r->query_get("dtls");

srs_trace("RTC data %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, srtp=%s, dtls=%s",
streamurl.c_str(), api.c_str(), clientip.c_str(), ruc.req_->app.c_str(), ruc.req_->stream.c_str(), remote_sdp_str.length(), eip.c_str(),
srtp.c_str(), dtls.c_str());

ruc.eip_ = eip;
ruc.publish_ = false;
ruc.dtls_ = (dtls != "false");

if (srtp.empty()) {
ruc.srtp_ = _srs_config->get_rtc_server_encrypt();
} else {
ruc.srtp_ = (srtp != "false");
}

// TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
if ((err = ruc.remote_sdp_.parse(remote_sdp_str)) != srs_success) {
return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
}

if ((err = check_remote_sdp(ruc.remote_sdp_)) != srs_success) {
return srs_error_wrap(err, "remote sdp check failed");
}

SrsSdp local_sdp;

// Config for SDP and session.
local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost);
local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost);

if ((err = exchange_sdp(ruc.req_, ruc.remote_sdp_, local_sdp)) != srs_success) {
return srs_error_wrap(err, "remote sdp have error or unsupport attributes");
}

// Whether enabled.
bool server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(ruc.req_->vhost);
if (server_enabled && !rtc_enabled) {
srs_warn("RTC disabled in vhost %s", ruc.req_->vhost.c_str());
}
if (!server_enabled || !rtc_enabled) {
return srs_error_new(ERROR_RTC_DISABLED, "Disabled server=%d, rtc=%d, vhost=%s",
server_enabled, rtc_enabled, ruc.req_->vhost.c_str());
}

// TODO: FIXME: When server enabled, but vhost disabled, should report error.
SrsRtcConnection* session = NULL;
if ((err = server_->create_session(&ruc, local_sdp, &session)) != srs_success) {
return srs_error_wrap(err, "create session");
}

ostringstream os;
if ((err = local_sdp.encode(os)) != srs_success) {
return srs_error_wrap(err, "encode sdp");
}

string local_sdp_str = os.str();

srs_verbose("local_sdp=%s", local_sdp_str.c_str());

res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str()));

// TODO: add candidates in response json?

res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
res->set("sessionid", SrsJsonAny::str(session->username().c_str()));

srs_trace("RTC data username=%s, offer=%dB, answer=%dB", session->username().c_str(),
remote_sdp_str.length(), local_sdp_str.length());

return err;
}

srs_error_t SrsGoApiRtcDataChannel::check_remote_sdp(const SrsSdp& remote_sdp)
{
srs_error_t err = srs_success;

if (remote_sdp.group_policy_ != "BUNDLE") {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "now only support BUNDLE, group policy=%s", remote_sdp.group_policy_.c_str());
}

if (remote_sdp.media_descs_.empty()) {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no media descriptions");
}

for (std::vector<SrsMediaDesc>::const_iterator iter = remote_sdp.media_descs_.begin(); iter != remote_sdp.media_descs_.end(); ++iter) {
if (iter->type_ != "application") {
return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "unsupport media type=%s", iter->type_.c_str());
}
}

return err;
}

srs_error_t SrsGoApiRtcDataChannel::exchange_sdp(SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp)
{
srs_error_t err = srs_success;
local_sdp.version_ = "0";

local_sdp.username_ = RTMP_SIG_SRS_SERVER;
local_sdp.session_id_ = srs_int2str((int64_t)this);
local_sdp.session_version_ = "2";
local_sdp.nettype_ = "IN";
local_sdp.addrtype_ = "IP4";
local_sdp.unicast_address_ = "0.0.0.0";

local_sdp.session_name_ = "SRSDataSession";

local_sdp.msid_semantic_ = "WMS";
local_sdp.msids_.push_back(req->app + "/" + req->stream);

local_sdp.group_policy_ = "BUNDLE";

for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_[i];
if (!remote_media_desc.is_application()) {
continue;
}

local_sdp.media_descs_.push_back(SrsMediaDesc("application"));

SrsMediaDesc& local_media_desc = local_sdp.media_descs_.back();

local_media_desc.mid_ = remote_media_desc.mid_;
local_sdp.groups_.push_back(local_media_desc.mid_);

local_media_desc.port_ = 9;
local_media_desc.protos_ = "UDP/DTLS/SCTP";

if (remote_media_desc.session_info_.setup_ == "active") {
local_media_desc.session_info_.setup_ = "passive";
} else if (remote_media_desc.session_info_.setup_ == "passive") {
local_media_desc.session_info_.setup_ = "active";
} else if (remote_media_desc.session_info_.setup_ == "actpass") {
local_media_desc.session_info_.setup_ = "passive";
} else {
// @see: https://tools.ietf.org/html/rfc4145#section-4.1
// The default value of the setup attribute in an offer/answer exchange
// is 'active' in the offer and 'passive' in the answer.
local_media_desc.session_info_.setup_ = "passive";
}

local_media_desc.rtcp_mux_ = true;
local_media_desc.rtcp_rsize_ = true;
}

return err;
}
#endif

SrsGoApiRtcPublish::SrsGoApiRtcPublish(SrsRtcServer* server)
{
server_ = server;
Expand Down
Loading

0 comments on commit e511fd9

Please sign in to comment.