This content has been marked as final. Show 7 replies
user11007219 wrote:RTP is designed for real-time streams, so maintaining real-time status is more important than quality...
the data drop by jitter buff at least >99%, why?
How can I improve this ?Well, you can make the internet less sucky about losing / delaying packets... transmit the RTP packets over TCP instead of UDP... or do the thing that you're already doing and increase the "Jitter Buf" to 500MS.
attar wrote:How the hell should I know, you said you'd already done it!
How to increase the "Jitter Buff" with JMF? RTPManager?
There is an example of changing around the capture and render buffers...
Additionally, there are a number of other "controls" you can get on your processor
I don't see anything specifically setup to allow you to change the jitter buffer size... only control I see that has any affect on RTP is the PacketSizeControl, which you could use. Smaller packets = more jitter, if I recall correctly.
I found the article about AudioBuffer, I fellow the step and add buff in my player(it is uesed to play audio from cilent-side rtp data) and datasource(it is uesed to get the micphone voice to transmit to cilent-side as rtp data), but it dose not worked...
so, could you have anothor solution?