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At 11:20 AM 4/18/2002 -0500, Ken Yeo wrote:
No, that won't happen. Much real-time voice and video traffic is constant bit-rate, though there are CODECs that offer variable bit rates within an envelope of min/max bit rates. When packet loss is encountered, the equipment will usually try error concealment to deal with the lost data -- how sophisticated this is depends on the equipment. Re-sending the lost data is pointless because it will arrive too late.-For UDP based audio/video trafffics, if the applications use RTSP and H.323, RTCP/H.245 will signal the sender to slowdown the transmission if the receiver lost packets.
More importantly, the sender will not slow down its transmission. It can't -- at any instant, there's a fixed amount of bandwidth required to carry the real-time voice and video, and there's no way to magically use less bandwidth. Buffering options are limited because the stream is real time and end-to-end latency must be bounded. And if the required bandwidth of the stream is greater than the available bandwidth over a long period of time (long in this case means seconds), no amount of buffering can help you. You need to abandon real-time and switch to a store-and-forward paradigm.
For example, a G.729a encoded real-time voice call requires 8Kbit/sec constant bit rate (excluding IP and RTP packet headers). If it can't get 8Kbit/sec, the user gets crappy voice quality.
Some streaming systems (non real-time -- the content is stored for on-demand viewing) encode the video/audio at several different data rates and try to guess the available bandwidth for a customer's connection. This process typically happens at the start of the streaming session, However, automatically switching to a lower rate later is hard to do, so it's rare to see it implemented. But none of this can be done for real-time voice or video.
H.323 is all about call signalling. It doesn't have anything to do with congestion management. Also, I think you mean H.323 (optionally including H.245) paired up with RTP -- this is the usual combination for VoIP calls. RTSP is typically used for streamed content.Did I miss anything? How about UDP traffics that are not using RTSP/H.323?
Other real-time UDP protocols (protocols used by games such as Half Life, for example) typically use the same kind of concepts and techniques that are in RTSP to detect packet loss so that they can conceal it or react in some other way. But if they're carrying a real-time service that has minimum bandwidth requirements, then there's no way for the sender to slow down.
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