Send mpeg-ts file source to SRT. Error about payload

Daniel Rossi electroteque at
Fri Jul 26 16:37:33 UTC 2019

according to gst-inspect-1.0 filesrc

I have a blocksize option. Do I set this to 1316?

inspecting tsparse doesn't say much, including command usage.

gst-launch-1.0 -v filesrc location=sintel_lang.ts blocksize=1316 ! 
tsparse ! srtsink uri=srt://:8888/

and the receiver

gst-launch-1.0 srtsrc uri=srt:// ! decodebin ! 

I get these errors.

02:34:56.655957*E: SRT.c: LiveSmoother: payload size: 18800 exceeds 
maximum allowed 1316

------ Original Message ------
From: "Nicolas Dufresne" <nicolas at>
To: "Daniel Rossi" <electroteque at>
Cc: "Discussion of the development of and with GStreamer" 
<gstreamer-devel at>
Sent: 7/26/2019 10:03:13 PM
Subject: Re: Re[2]: Send mpeg-ts file source to SRT. Error about payload

>Le jeu. 25 juill. 2019 23 h 30, Daniel Rossi <electroteque at> a 
>écrit :
>>There is an element called tsparse, but same thing.
>>gst-launch-1.0 -v filesrc location =sintel_lang.ts ! tsparse ! srtsink 
>You should use gst-inspect-1.0 to learn about the configuration for 
>filesrc and tsparse (I'm typing this from memory, and there exist in 
>usage of mpegts and ts as element name prefix). File source has an 
>option to configure the read size, these needs to be multiple of 188 
>and max to 1316. The ts parse as an option to add and smooth timestamp, 
>these need to be configured.
>>my pullside for the test is
>>gst-launch-1.0 srtsrc uri=srt:// ! decodebin ! 
>>------ Original Message ------
>>From: "Nicolas Dufresne" <nicolas at>
>>To: "Daniel Rossi" <electroteque at>; "Discussion of the 
>>development of and with GStreamer" 
>><gstreamer-devel at>
>>Sent: 7/26/2019 1:19:54 PM
>>Subject: Re: Send mpeg-ts file source to SRT. Error about payload
>>>Le jeu. 25 juill. 2019 22 h 25, Daniel Rossi <electroteque at> 
>>>a écrit :
>>>>I'm trying to send an mpeg-ts source over SRT for multi language 
>>>>track testing.
>>>>When pulling this stream I am getting an internal error.
>>>>gst-launch-1.0 -v filesrc location =sintel_lang.ts !  rtpstreampay ! 
>>>>srtsink uri=srt://:8888/
>>>>Setting pipeline to PAUSED ...
>>>>Pipeline is PREROLLING ...
>>>>Pipeline is PREROLLED ...
>>>>Setting pipeline to PLAYING ...
>>>>New clock: GstSystemClock
>>>>12:13:33.532337/filesrc0:src*E: SRT.c: LiveSmoother: payload size: 
>>>>4098 exceeds maximum allowed 1316
>>>a) why do you use stream pay ?
>>>b) you might want to use mpegtsparse to timestamp your stream
>>>c) configure filesrc to read 1316 bytes to fix this error.
>>>>Is there also a way to pipeline a h264 file with multi audio tracks 
>>>>through mpeg-ts  and set PID numbers for each track instead of 
>>>>outputting to TS via ffmpeg first ?
>>>>gstreamer-devel mailing list
>>>>gstreamer-devel at
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