srs-5.0.12 uses Hikvision camera GB28181 for normal RTMP playback, but cannot play WebRTC.

[kirofreedom]

Note: Issues that do not provide the following information will be directly deleted (Please follow the issue template, or we will delete it) ' Make sure to maintain the markdown structure.

Note: For inquiries and discussions, please submit them to the SRS community (Please ask questions at) http://bbs.ossrs.net ' Make sure to maintain the markdown structure.

Description

Please describe your issue here.

1. SRS version: 5.0.12 (feature/gb28181 branch)

2. The log of SRS is as follows:

   [2021-09-29 18:10:36.005][Warn][128032][39r5083u][11] RTC error code=5018 : create session, dtls=1, srtp=1, eip= : create session : add player : play negotiate : no valid found h264 payload type
   thread [128032][39r5083u]: do_serve_http() [src/app/srs_app_rtc_api.cpp:191][errno=11]
   thread [128032][39r5083u]: create_session() [src/app/srs_app_rtc_server.cpp:480][errno=11]
   thread [128032][39r5083u]: do_create_session() [src/app/srs_app_rtc_server.cpp:501][errno=11]
   thread [128032][39r5083u]: add_player() [src/app/srs_app_rtc_conn.cpp:2020][errno=11]
   thread [128032][39r5083u]: negotiate_play_capability() [src/app/srs_app_rtc_conn.cpp:3196][errno=11]
   [2021-09-29 18:10:36.047][Trace][128032][39r5083u] TCP: before dispose resource(HttpConn)(0x22ed690), conns=4, zombies=0, ign=0, inz=0, ind=0
   [2021-09-29 18:10:36.047][Warn][128032][39r5083u][104] client disconnect peer. ret=1007
   [2021-09-29 18:10:36.047][Trace][128032][0py0r282] TCP: clear zombies=1 resources, conns=4, removing=0, unsubs=0
   [2021-09-29 18:10:36.047][Trace][128032][39r5083u] TCP: disposing #0 resource(HttpConn)(0x22ed690), conns=4, disposing=1, zombies=0
   [2021-09-29 18:10:36.787][Trace][128032][0x84e94z] gb28181: client id=34020000001320000001@34020000001320000001,  ssrc=0x8ac009, peer(192.168.2.14, 15060), rtmp muxer is alive

3. The configuration of SRS is as follows:

   
   # push gb28181 stream to SRS.

listen 1935; max_connections 1000; daemon off; srs_log_tank console;

http_api { enabled on; listen 1985; }

http_server { enabled on; listen 8080; }

stats { network 0; }

stream_caster { enabled on; caster gb28181;

Forward the stream to the rtmp server address and port

TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104

[stream] is the VideoChannelCodecID for sip

The automatically created channel [stream] is 'chid[ssrc]' where [ssrc] is the rtp ssrc

[ssrc] is the ssrc in rtp

output rtmp://127.0.0.1:1935/live/[stream];

Multiplexing port for receiving rtp streams from the receiving device side

listen 9000;

Multiplexing port type, on for tcp, off for udp

Default: on

tcp_enable on;

Range of RTP receiving listening ports, minimum value

rtp_port_min 58200;

Range of RTP receiving listening ports, maximum value

rtp_port_max 58300;

Whether to wait for a keyframe before forwarding,

off: No need to wait, forward directly

on: Wait for the first keyframe before forwarding

wait_keyframe on;

Idle waiting time for RTP packets, if no packets are received within the specified time

the RTP listening connection will automatically stop and send a BYE command.

rtp_idle_timeout 30;

Whether to forward the audio stream

Currently only supports AAC format, so the device needs to support AAC format

on: forward audio

off: do not forward audio, only video

Note!!!: flv only supports three sample rates: 11025, 22050, 44100

If the device does not support any of these three, it will automatically select one format during forwarding

It will also encapsulate the ADTS header in the FLV AAC raw data

This way, the player will automatically select the sampling frequency based on the ADTS header

Players like ffplay and vlc can do this, but flash does not have sound

because flash only supports 11025, 22050, 44100.

audio_enable off;

Whether to enable RTP buffering

Enabling it can effectively solve issues like RTP out-of-order

It is recommended to disable it for TCP mode.

jitterbuffer_enable off;

Server host, can be a domain name or IP address

This is the address to which the device sends media. If the server is on an internal or external network,

the external address needs to be specified.

The IP address returned by the API when creating a stream session is also the host.

$CANDIDATE is a system environment variable used to obtain the address from the environment variables.

If it is not configured, "*" is used to specify the network card address for stats network.

If the network is not configured, the default is the address of the 0th network card.

TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594

host $CANDIDATE;

host 192.168.2.200;

Create an RTMP media channel based on the received PS RTP packets, no need to create it through the API interface.

The RTMP address parameter [stream] is the channel ID in the format chid[ssrc].

auto_create_channel off;

sip {

   # Whether to enable internal SIP signaling in SRS.

Use "on" to route signaling through SRS, and "off" to only forward PS streams.

enabled on;

   # UDP port for SIP listening.

listen 5060;

   # SIP server ID.

The device configuration number needs to be consistent with this value, otherwise registration will fail.

serial 34020000002000000001;

   # SIP server domain.

    realm               3402000000;

   # Timeout for receiving response after sending ACK from the server, in seconds.

If no response is received within the specified time, it is considered a failure.

ack_timeout 30;

   # Device heartbeat maintenance time, if no heartbeat is received within the specified time (in seconds),

the device is considered offline.

keepalive_timeout 120;

   # Whether to automatically send an invite to the device after registration

on: yes off: no, needs to be controlled through API.

auto_play on;

Whether the port for sending streams from the device is fixed

on: send streams to a multiplexing port like 9000

off: automatically select a port from the range between rtp_mix_port and rtp_max_port that is available.

invite_port_fixed on;

   # Interval for querying the device list from the device or subdomain, in seconds

Default is 60 seconds

query_catalog_interval 60; } }

rtc_server { enabled on;

Listen at udp://8000

listen          8000;
#
# The $CANDIDATE means fetch from env, if not configed, use * as default.
#
# The * means retrieving server IP automatically, from all network interfaces,
# @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
# candidate       $CANDIDATE;
candidate       192.168.2.200;

}

vhost defaultVhost { rtc { enabled on; bframe discard; }

http_remux {
    enabled     on;
    mount       [vhost]/[app]/[stream].flv;
}

}

**Replay**

**How to replay bug?**

> Steps to reproduce the bug

**How to replay bug?**

1. Start the Hikvision camera (gb28181).
2. Run the srs_gb28181.html webpage.
3. Select "GB28181" and query the GB28181 media channel.
4. After selecting an existing channel, click on "RTC Play". An error occurs, but playing RTMP stream using VLC works fine.

**Expected behavior**

The expectation is to be able to play WebRTC streams normally.

`TRANS_BY_GPT3`

[winlinvip]

This error is very clear: no valid found h264 payload type.

TRANS_BY_GPT3

[kirofreedom]

Hmm, the backend display shows that there is a stream, and rtmp can play normally. The camera uses h264 encoding. I am currently testing with srs. I am just starting to work on audio and video, and I am still a beginner. I appreciate the help from all the experts.

TRANS_BY_GPT3

[qichaoshen82]

Suggest using the latest gb28181 branch code and adding the "rtmp_to_rtc on;" field in the configuration file.

If there are still issues after using the latest code and adding "rtmp_to_rtc on;", you can reopen the issue. This issue can be closed as per the suggestion. @duiniuluantanqin @winlinvip

TRANS_BY_GPT3

[tifcty1]

The latest version of GB seems to indeed have issues with RTC playback, unlike previous versions like 5.0.12. Enabling 'rtmp_to_rtc on' resolves the issue, but it is unclear whether it is a bug or not.

It is recommended to use the latest gb28181 branch code and add the 'rtmp_to_rtc on' field in the configuration file.

If there are still issues after using the latest code and adding "rtmp_to_rtc on;", you can reopen the issue. This issue is recommended to be closed. @duiniuluantanqin @winlinvip

TRANS_BY_GPT3

[duiniuluantanqin]

Previously, rtmp_to_rtc was enabled by default, but now it is off by default. This is not a bug.

TRANS_BY_GPT3

[duiniuluantanqin]

@kirofreedom https://github.com/ossrs/srs/issues/2642#issuecomment-971080217

[tifcty1]

So that's how it is, thank you 🙏

TRANS_BY_GPT3

[kirofreedom]

It is recommended to use the latest gb28181 branch code and add the "rtmp_to_rtc on;" field in the configuration file.

If there are still issues after using the latest code and adding "rtmp_to_rtc on;", you can reopen the issue. This issue is suggested to be closed. @duiniuluantanqin @winlinvip

Thank you very much!! I will give it a try.

TRANS_BY_GPT3

[hhw3KevinHou]

rtmp_to_rtc on;

Good job, I have been working on it for more than a day. I searched through numerous online articles but couldn't find a reliable answer. Only here did I find the solution. I love you so much.

TRANS_BY_GPT3

[winlinvip]

Here, https://github.com/ossrs/srs/issues/2716#webrtc-live, it explains in detail why it is set to default off. Originally, default on would be more convenient, but there were too many issues, so it had to be changed to off.

If you have any issues, please search in the official repository. The information found online is mostly copied and may not be up to date.

Then I will close this issue. Thank you all.

TRANS_BY_GPT3