[Live-devel] How to set Presentation Time correctly for audio frame in rtsp server?

[Ross Finlayson]

> I try on many ways but still something freezing my audio, Its like one second audio and two not. I want to be sure I am doing this properly:
>  
> I am sending audio as rtsp server, its PCMU/8000, I give it every second frame sized: 8000 bytes and send it.

That’s much too large.  First, the maximum size of RTP packets is, by default, only about 1500 bytes.  Second, even if you were to reconfigure your server code to support 8000-byte RTP packets, it’s not clear whether or not packets that large would be able to get transmitted - without IP fragmentation - to the server.  Finally, if this 8000 byte packet were to get lost (which is very likely if you have IP-level fragmentation), then you’d lose the whole 1 second of audio.

It’d be much better to (for example) grab 1/10 second worth of audio at a time - i.e., 800 bytes worth, setting “fDurationInMicroseconds” to 100000.

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

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