I have to send a sequence of video frames over UDP as fast and real-time as possible and while I got the basics working, I am running into all sorts of difficulties. Some of my goals:
我必须在UDP上尽可能快和实时地发送一系列视频帧，在基本工作的同时，我遇到了各种各样的困难。我的一些目标:
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Data will normally be sent over dial-up (hence UDP instead of TCP), but also needs to support fast Ethernet.
数据通常通过拨号(因此UDP而不是TCP)发送，但也需要支持快速以太网。
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It's OK to occasionally drop frames (hence UDP instead of TCP).
偶尔删除帧是可以的(因此UDP而不是TCP)。
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Need low latency. The frame the remote receives should be one that was recently sent (no more than a few frames waiting in buffers).
需要低延迟。远程接收的帧应该是最近发送的(不超过几个帧在缓冲区中等待)。
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I need to be able to detect the effective bandwidth so that I can compress the frames more or less to keep frame rate up.
我需要能够检测有效的带宽，这样我就可以或多或少地压缩帧来保持帧速率。
I have managed to implement most of the pieces:
我成功地实现了大部分的部分:
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I break up frame data into one or more datagrams of about 500 bytes and each has a sequence number and other info. The receiver reassembles the entire frame and detects if any datagrams are missing.
我将帧数据分解为一个或多个数据报，大约500字节，每个数据报都有序列号和其他信息。接收器重新组装整个帧并检测是否有任何数据报丢失。
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If the receiver detects more than a certain percentage of dropped frames (e.g. 50% over the last 10 frames), I send a TCP message to the sender to slow down by 50%. Sender than slowly increases speed by 5% each subsequent frame.
如果接收方检测到超过一定百分比的丢失帧(例如，在最后10帧中50%)，我将向发送方发送TCP消息，以降低50%的速度。发送者比缓慢增加速度5%每后续帧。
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Using System.Net.Sockets.UdpClient to send and receive the data.
使用System.Net.Sockets。UdpClient用于发送和接收数据。
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I have a separate TCP channel used for control messages back to sender.
我有一个单独的TCP通道，用于控制返回给发送者的消息。
My main difficulty right now is detecting the effective bandwidth and dealing with latency, especially over dial-up (max ~4,000 bytes/sec). For example, if I try to send 100,000 bytes/second using TcpClient.Send() they ALL seem to arrive (no dropped datagrams) but with large latency by the time last datagram arrives. I think the TcpClient.Send() function is blocking until the buffer is able to send which messes up my current algorithm.
我现在的主要困难是检测有效的带宽和处理延迟，特别是拨号上网(最大4000字节/秒)。例如，如果我尝试使用TcpClient.Send()发送100,000字节/秒，它们似乎都到达了(没有丢失的数据报)，但是在最后一个数据报到达时，延迟会很大。我认为TcpClient.Send()函数正在阻塞，直到缓冲区能够发送，这就打乱了我当前的算法。
Can anybody point me to any sources of information for how to:
谁能告诉我如何:
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Detect actual bandwidth over UDP.
检测UDP的实际带宽。
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A better algorithm for dynamically adjusting bandwidth to suit the available pipe.
一种更好的动态调整带宽以适应可用管道的算法。
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Send data smoothly at the desired bandwidth.
在所需的带宽上平稳地发送数据。
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A way to detect and keep latency down to a minimum.
一种将延迟降低到最小的方法。
I have been spinning my wheels over the last week and every time I solve one problem it seems another rears up is head.
在过去的一周里，我一直在不停地思考，每次我解决了一个问题，就好像另一个问题又出现了。
1 个解决方案
#1
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You can also to add a timestamp to every packet. Then you can detect if the delay increase. In this case you send back a message to reduce the bandwidth.
您还可以向每个包添加时间戳。然后你可以检测延迟是否增加。在这种情况下，您发送一个消息以减少带宽。
On creating the connection you detect the latency with very few packets. This value should not change at running.
在创建连接时，您可以用非常少的数据包来检测延迟。此值在运行时不应更改。
#1
2
You can also to add a timestamp to every packet. Then you can detect if the delay increase. In this case you send back a message to reduce the bandwidth.
您还可以向每个包添加时间戳。然后你可以检测延迟是否增加。在这种情况下，您发送一个消息以减少带宽。
On creating the connection you detect the latency with very few packets. This value should not change at running.
在创建连接时，您可以用非常少的数据包来检测延迟。此值在运行时不应更改。