Dear all. Pardon the ignorance, but I'm wondering about the relationship between RTP and the (for example) Speex jitter buffer. This may not relate directly to speex as such, but I hope you'll take the time to reply anyway should you have any details for me. If I would implement my VoIP application using RTP to transmit data between recording-point and playback-point, would I still use the jitter buffer or does RTP actually take care of the jitter buffer tasks? If I use the Speex jitter buffer implementation (or any other jitter buffer implementation for that matter), should I expect to get a decent voice data transfer/decoding by sending the speex encoded data (along with a simple time-stamp of when the frame/packet was prepared) via UDP and relying on the jitter buffer implementation to supply a steady stream of ready-to-use packets? Respectfully, Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20050606/0cab0fec/attachment.htm