RootEncoder: RTSP - MISSING NAL - WOWZA Streaming engine

Hi, @pedroSG94 , I really appreciate your work on this library, especially maintenance and support you do here!

I’m working on an application and used this library to do RTMP live streaming toward a wowza streaming engine. Recently I started to switch to try RTSP as it may encode into h265 and server should support it fine.

At the end, after I setup and rtspclient I’m facing an issue, when I start streaming to the server URL. On the server side I see warnings like: RTPTrack.getCodecConfig(video): Missing NAL SPS(7) , RTPTrack.getCodecConfig(video): Missing NAL PPS(8) then RTPTrack.checkRTCPSSRC[RTPStream={streamcontext=livestream/_definst_/myStream,mode=PUBLISH,uuid=925522951},RTPTrack={ourSsrc=1151399308/audio}]: ssrc error: expected:da45a80e, got:da6a30d6 from host:null

and RTPTrack.checkRTCPSSRC[RTPStream={streamcontext=livestream/_definst_/myStream,mode=PUBLISH,uuid=925522951},RTPTrack={ourSsrc=1051487197/video}]: ssrc error: expected:cf3d0aac, got:c51b90a6 from host:null

from android client I can see

I/OpenGlViewBase: Thread started. I/AudioEncoder: AudioEncoder started I/MicrophoneManager: Microphone started I/RtspClient: OPTIONS rtsp://my.domain.com:1935/livestream/myStream RTSP/1.0 CSeq: 1 I/RtspClient: RTSP/1.0 200 OK CSeq: 1 Server: Wowza Streaming Engine 4.8.13+1 build20210527172944 Cache-Control: no-cache Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS, ANNOUNCE, RECORD, GET_PARAMETER Supported: play.basic, con.persistent ANNOUNCE rtsp://my.domain.com:1935/livestream/myStream RTSP/1.0 CSeq: 2 Content-Length: 448 Content-Type: application/sdp

v=0
o=- 0 0 IN IP4 127.0.0.1
s=Unnamed
i=N/A
c=IN IP4 feeder.flow.tours
t=0 0
a=recvonly
m=video 0 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;sprop-parameter-sets=WjBLQUh0b0hnVVpB,YU00TmlBPT0=;
a=control:trackID=1
m=audio 0 RTP/AVP 96
a=rtpmap:96 MPEG4-GENERIC/32000/2
a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=1290; SizeLength=13; IndexLength=3; IndexDeltaLength=3;
a=control:trackID=0

I/RtspClient: RTSP/1.0 200 OK CSeq: 2 Server: Wowza Streaming Engine 4.8.13+1 build20210527172944 Cache-Control: no-cache Session: 1428659075;timeout=60 SETUP rtsp://my.domain.com:1935/livestream/myStream/trackID=0 RTSP/1.0 Transport: RTP/AVP/TCP;interleaved=0-1;mode=record CSeq: 3 Session: 1428659075 I/RtspClient: RTSP/1.0 200 OK CSeq: 3 Server: Wowza Streaming Engine 4.8.13+1 build20210527172944 Cache-Control: no-cache Expires: Tue, 2 Nov 2021 22:38:04 UTC Transport: RTP/AVP/TCP;interleaved=0-1;mode=record Date: Tue, 2 Nov 2021 22:38:04 UTC Session: 1428659075;timeout=60 SETUP rtsp://my.domain.com:1935/livestream/myStream/trackID=1 RTSP/1.0 Transport: RTP/AVP/TCP;interleaved=2-3;mode=record CSeq: 4 Session: 1428659075 I/RtspClient: RTSP/1.0 200 OK CSeq: 4 Server: Wowza Streaming Engine 4.8.13+1 build20210527172944 Cache-Control: no-cache Expires: Tue, 2 Nov 2021 22:38:04 UTC Transport: RTP/AVP/TCP;interleaved=2-3;mode=record Date: Tue, 2 Nov 2021 22:38:04 UTC Session: 1428659075;timeout=60 I/RtspClient: RECORD rtsp://my.domain.com:1935/livestream/myStream RTSP/1.0 Range: npt=0.000- CSeq: 5 Session: 1428659075 I/RtspClient: RTSP/1.0 200 OK CSeq: 5 Server: Wowza Streaming Engine 4.8.13+1 build20210527172944 Cache-Control: no-cache Range: npt=now- Session: 1428659075;timeout=60 I/BaseRtpSocket: wrote packet: Audio, size: 381 I/BaseSenderReport: wrote report: Audio, packets: 1, octet: 381 I/MainActivity: at tours.flow.hdrstreaming(null:-1) . [run] — Rtmp サーバーへ 接続成功しました。 D/ACodec: dataspace changed to 0x10c60000 (R:2(Limited), P:3(BT601_6_625), M:3(BT601_6), T:3(SMPTE170M)) (R:2(Limited), S:6(BT2020), T:3(SMPTE_170M)) I/BaseRtpSocket: wrote packet: Audio, size: 382 I/BaseRtpSocket: wrote packet: Audio, size: 391 I/BaseRtpSocket: wrote packet: Audio, size: 391 I/BaseRtpSocket: wrote packet: Audio, size: 392 I/BaseRtpSocket: wrote packet: Video, size: 38 I/BaseSenderReport: wrote report: Video, packets: 1, octet: 38 I/BaseRtpSocket: wrote packet: Audio, size: 398 I/BaseRtpSocket: wrote packet: Audio, size: 398 I/BaseRtpSocket: wrote packet: Audio, size: 400 wrote packet: Audio, size: 388 I/BaseRtpSocket: wrote packet: Video, size: 37 I/BaseRtpSocket: wrote packet: Video, size: 1272

I can’t see video when I try to play via player, which works fine when streaming to same server from tools for android on another device. This is an older android 7.1 device (sdk api is 25) and I’m using an older library of yours because of limitations of this device (it’s an android based Theta Z1 360 degree camera I’m playing with).

I’m using library 1.4.8, even upgrading to this version and make it work took some efforts because of compatibility issues I was facing with…

Thanks for your thoughts and help in advance!

Regards, Laszlo