RTC: Supports Firefox and mobile access. Also fixes some memory leak bugs.

[duiniuluantanqin]

Fix the bug that the new version does not support Firefox and mobile access to WebRTC.

Reason
The payload type in the RTP header is inconsistent with the one in the SDP.

Solution
Save the payload type negotiated during SDP negotiation in the track. When calling send_packets, update this PT in each RTP header.

Test Results

The italic bold line indicates the newly added support (compared to version 4.0.39).

1. PC端 (PC end)
   • Chrome v86.0.4240.75
   • Firefox v81.0.1
2. Android
   • Chrome v85
3. IOS
   • Safari v12

issue
Resolve the issues of the following 2 items.

• rtc playback, successful in Chrome, failed in Firefox. #1962
• https://github.com/ossrs/srs/issues/1966

Others

1. Fix the memory leak issue in the SrsRtcConnection::add_player function.
2. Fix the SDP exception issue in the generate_media_payload_type function when the first parameter is missing, such as a=fmtp 111 ;useinbandfec=1.
3. Add two new audio parameters: stereo and maxplaybackrate.

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TRANS_BY_GPT3

[winlinvip]

👍 I took a quick look and didn't see any issues. I will find time to examine the merge in detail. Thank you, thank you.

TRANS_BY_GPT3

[winlinvip]

I tried it in detail, the PT in Answer must be in Offer.

For RTMP, currently SRS defaults to Chrome's PT (Opus/111, H264/102), while Firefox (Opus/109, H264/126) is different.

For RTC, SRS generates the Player's PT based on the Publisher's PT. So, if Chrome is used for streaming and playback, there are no issues. Similarly, if Firefox is used for both streaming and playback, there are no problems (confirmed). However, when Chrome is used for streaming and Firefox for playback, or vice versa, there may be situations where packets are received but cannot be played.

Therefore, the solution to this problem should be: The PT of the Answer should be generated based on the Offer, it should not be set as default or use the PT of the Publisher.

TRANS_BY_GPT3

[ghostsf]

I tried it in detail, and the PT in Answer must be in Offer.

For RTMP, currently SRS defaults to Chrome's PT (Opus/111, H264/102), while Firefox's PT (Opus/109, H264/126) is different.

For RTC, SRS generates the Player's PT based on the Publisher's PT. So, if Chrome is used for publishing and playback, there are no issues. Similarly, if Firefox is used for both publishing and playback, there are no issues (confirmed to be working fine). However, when there is a cross between Chrome publishing and Firefox playback, or Firefox publishing and Chrome playback, there may be a situation where packets are received but cannot be played.

Therefore, the solution to this problem should be: the PT of the Answer should be generated based on the Offer, it should not be set as default or use the PT of the Publisher.

Yes, please merge and process it quickly.

TRANS_BY_GPT3

[winlinvip]

In addition, fmtp is also necessary, otherwise decoding will fail.

a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1

Note: H.264 is necessary, but for Opus, further confirmation is required.

TRANS_BY_GPT3

[winlinvip]

Firefox still has an issue where the Candidate cannot be configured as a domain name, it must be configured as an IP address.

• Firefox works fine: http://ossrs.net/players/rtc_player.html?vhost=d.ossrs.net&server=d.ossrs.net&api=11985&autostart=true&target=39.107.238.185
• Firefox cannot play when the Candidate is a domain name: http://ossrs.net/players/rtc_player.html?vhost=d.ossrs.net&server=d.ossrs.net&api=11985&autostart=true&target=d.ossrs.net

You can change the domain name to an IP address in the configuration.

    rtc_server {
      enabled         on;
      listen          8000;
      # Firefox doesn't work with domain.
      #candidate       d.ossrs.net;
      candidate       39.107.238.185;
    }

TRANS_BY_GPT3

[duiniuluantanqin]

Firefox still has an issue, Candidate cannot be configured as a domain name, it must be configured as an IP address:

• Firefox works fine: http://ossrs.net/players/rtc_player.html?vhost=d.ossrs.net&server=d.ossrs.net&api=11985&autostart=true&target=39.107.238.185
• Firefox cannot play when Candidate is a domain name: http://ossrs.net/players/rtc_player.html?vhost=d.ossrs.net&server=d.ossrs.net&api=11985&autostart=true&target=d.ossrs.net

You can change the domain name to an IP address in the configuration.

    rtc_server {
      enabled         on;
      listen          8000;
      # Firefox doesn't work with domain.
      #candidate       d.ossrs.net;
      candidate       39.107.238.185;
    }

The approach to solving this problem is what? Can someone give me some guidance? I will give it a try.

TRANS_BY_GPT3

[duiniuluantanqin]

This line of code is used to remove the first semicolon. The purpose of having the first semicolon is to avoid formatting errors that may occur when the first parameter is missing.

TRANS_BY_GPT3

[ghostsf]

yes