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Code Issues Releases Wiki Activity

Squash: Fix padding packets for RTMP2RTC

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This commit is contained in:
winlin 2021-07-08 14:37:18 +08:00
parent c8a1e0f3da
commit 8f91a90f28
6 changed files with 18 additions and 8 deletions
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1
CHANGELOG.md
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@ -20,6 +20,7 @@ The changelog for SRS.
## SRS 4.0 Changelog ## SRS 4.0 Changelog
* v4.0, 2021-07-08, For [#2403](https://github.com/ossrs/srs/issues/2403), fix padding packets for RTMP2RTC. 4.0.140
* v4.0, 2021-07-04, For [#2424](https://github.com/ossrs/srs/issues/2424), use srandom/random to generate. 4.0.139 * v4.0, 2021-07-04, For [#2424](https://github.com/ossrs/srs/issues/2424), use srandom/random to generate. 4.0.139
* v4.0, 2021-07-01, Merge [#2452](https://github.com/ossrs/srs/pull/2452), fix FFmpeg bug by updating channel_layout. 4.0.138 * v4.0, 2021-07-01, Merge [#2452](https://github.com/ossrs/srs/pull/2452), fix FFmpeg bug by updating channel_layout. 4.0.138
* 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 * 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

2
README.md
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@ -22,7 +22,7 @@ please set the CANDIDATE ([CN][v4_CN_WebRTC#config-candidate],[EN][v4_EN_WebRTC#
```bash ```bash
docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \ docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \
--env CANDIDATE=$(ifconfig en0 inet| grep 'inet '|awk '{print $2}') -p 8000:8000/udp \ --env CANDIDATE=$(ifconfig en0 inet| grep 'inet '|awk '{print $2}') -p 8000:8000/udp \
ossrs/srs:v4.0.117 ./objs/srs -c conf/srs.conf ossrs/srs:v4.0.139 ./objs/srs -c conf/srs.conf
``` ```
<a name="usage-source"></a> <a name="usage-source"></a>

8
trunk/research/players/js/srs.sdk.js
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@ -33,8 +33,8 @@ function SrsRtcPublisherAsync() {
// webrtc://r.ossrs.net:11985/live/mystream // webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com: // or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream // webrtc://myapi.domain.com/live/livestream
// or set the candidate(ip) of answer: // or set the candidate(eip) of answer:
// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185 // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
// or force to access https API: // or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https // webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP: // or use plaintext, without SRTP:
@ -272,8 +272,8 @@ function SrsRtcPlayerAsync() {
// webrtc://r.ossrs.net:11985/live/mystream // webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com: // or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream // webrtc://myapi.domain.com/live/livestream
// or set the candidate(ip) of answer: // or set the candidate(eip) of answer:
// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185 // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
// or force to access https API: // or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https // webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP: // or use plaintext, without SRTP:

10
trunk/src/app/srs_app_rtc_api.cpp
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@ -118,8 +118,11 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param); srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
} }
// For client to specifies the EIP of server. // For client to specifies the candidate(EIP) of server.
string eip = r->query_get("eip"); string eip = r->query_get("eip");
if (eip.empty()) {
eip = r->query_get("candidate");
}
string codec = r->query_get("codec"); string codec = r->query_get("codec");
// For client to specifies whether encrypt by SRTP. // For client to specifies whether encrypt by SRTP.
string srtp = r->query_get("encrypt"); string srtp = r->query_get("encrypt");
@ -338,8 +341,11 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param); srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
} }
// For client to specifies the EIP of server. // For client to specifies the candidate(EIP) of server.
string eip = r->query_get("eip"); string eip = r->query_get("eip");
if (eip.empty()) {
eip = r->query_get("candidate");
}
string codec = r->query_get("codec"); string codec = r->query_get("codec");
srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s", srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",

3
trunk/src/app/srs_app_rtc_sdp.cpp
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@ -747,6 +747,9 @@ srs_error_t SrsSdp::parse(const std::string& sdp_str)
line.erase(line.size()-1, 1); line.erase(line.size()-1, 1);
} }
// Strip the space of line, for pion WebRTC client.
line = srs_string_trim_end(line, " ");
if ((err = parse_line(line)) != srs_success) { if ((err = parse_line(line)) != srs_success) {
return srs_error_wrap(err, "parse sdp line failed"); return srs_error_wrap(err, "parse sdp line failed");
} }

2
trunk/src/core/srs_core_version4.hpp
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@ -9,6 +9,6 @@
#define VERSION_MAJOR 4 #define VERSION_MAJOR 4
#define VERSION_MINOR 0 #define VERSION_MINOR 0
#define VERSION_REVISION 139 #define VERSION_REVISION 140
#endif #endif