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Squash: Fix padding packets for RTMP2RTC
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6 changed files with 18 additions and 8 deletions
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@ -20,6 +20,7 @@ The changelog for SRS.
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## SRS 4.0 Changelog
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* v4.0, 2021-07-08, For [#2403](https://github.com/ossrs/srs/issues/2403), fix padding packets for RTMP2RTC. 4.0.140
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* v4.0, 2021-07-04, For [#2424](https://github.com/ossrs/srs/issues/2424), use srandom/random to generate. 4.0.139
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* v4.0, 2021-07-01, Merge [#2452](https://github.com/ossrs/srs/pull/2452), fix FFmpeg bug by updating channel_layout. 4.0.138
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* v4.0, 2021-06-30, Merge [#2440](https://github.com/ossrs/srs/pull/2440), fix [#2390](https://github.com/ossrs/srs/issues/2390), SRT bug for zerolatency. 4.0.137
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@ -22,7 +22,7 @@ please set the CANDIDATE ([CN][v4_CN_WebRTC#config-candidate],[EN][v4_EN_WebRTC#
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```bash
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docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \
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--env CANDIDATE=$(ifconfig en0 inet| grep 'inet '|awk '{print $2}') -p 8000:8000/udp \
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ossrs/srs:v4.0.117 ./objs/srs -c conf/srs.conf
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ossrs/srs:v4.0.139 ./objs/srs -c conf/srs.conf
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```
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<a name="usage-source"></a>
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@ -33,8 +33,8 @@ function SrsRtcPublisherAsync() {
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(ip) of answer:
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// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
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// or set the candidate(eip) of answer:
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// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
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// or use plaintext, without SRTP:
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@ -272,8 +272,8 @@ function SrsRtcPlayerAsync() {
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(ip) of answer:
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// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
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// or set the candidate(eip) of answer:
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// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
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// or use plaintext, without SRTP:
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@ -118,8 +118,11 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
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srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
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}
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// For client to specifies the EIP of server.
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// For client to specifies the candidate(EIP) of server.
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string eip = r->query_get("eip");
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if (eip.empty()) {
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eip = r->query_get("candidate");
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}
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string codec = r->query_get("codec");
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// For client to specifies whether encrypt by SRTP.
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string srtp = r->query_get("encrypt");
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@ -338,8 +341,11 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
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srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
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}
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// For client to specifies the EIP of server.
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// For client to specifies the candidate(EIP) of server.
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string eip = r->query_get("eip");
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if (eip.empty()) {
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eip = r->query_get("candidate");
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}
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string codec = r->query_get("codec");
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srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
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@ -747,6 +747,9 @@ srs_error_t SrsSdp::parse(const std::string& sdp_str)
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line.erase(line.size()-1, 1);
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}
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// Strip the space of line, for pion WebRTC client.
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line = srs_string_trim_end(line, " ");
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if ((err = parse_line(line)) != srs_success) {
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return srs_error_wrap(err, "parse sdp line failed");
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}
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@ -9,6 +9,6 @@
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#define VERSION_MAJOR 4
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#define VERSION_MINOR 0
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#define VERSION_REVISION 139
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#define VERSION_REVISION 140
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#endif
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