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Article ID: 000043898
Last Modified Date: 10/12/2021
Access Level: Public

What are the effects of network latency and jitter on an H.323 video call

Network Latency When using H.323 there are 4 separate data streams. Two half-duplex video streams and two half-duplex audio streams. This can be synopsized as one set video+audio data streams from each endpoint to the other. When discussing network latencies relative to the operation of H.323, there are 3 general categories to consider:
  • End-to-End latency in a given direction. This category addresses the total transit time for data of a given data stream to arrive at the remote endpoint. It is preferable for this transit time to not exceed 300 milliseconds. The average packet size of a video stream versus that of an audio stream is very different. Video packets sizes are usually large (800-1500 bytes) while audio packet sizes are generally small (480 bytes or less). This means that the average transit time for an audio stream can be less than that for a video stream if an intervening router or bridge prioritizes smaller over larger packets when encountering network congestion. Once can use "ping", with its packet size and flood options, and "tracert" as simple, convenient tools to determine average network transit times.
  • Intra-stream latency. This category addresses latencies within a given data stream which boils down to inter-packet latencies that deviate outside of the normal transmit time by more than 30-35 milliseconds, additional, in a 30 FPS stream, or 60 milliseconds in a 15 FPS stream. An example would be a data stream in a 30 FPS H.323 session that has an average transit time of 115 milliseconds. If a single packet in this stream encountered a transit time of 145 milliseconds or more (relative to a prior packet), it could cause a receive underrun condition at the receiving endpoint potentially causing either blocky, jittery video or undesirable audio artifacts. Also, intra-stream latencies can cause inter-stream latencies, which are discussed next.
  • Inter-stream latency. This category addresses the relative latencies that can be encountered between the audio and video data streams. This is where the relative average transit time for the given streams, at any given point, vary from each other. In this case the relative latency variations are not symmetrical. This is due to the fact that the human brain already compensates for audio latency relative to video. Due to this fact, an audio stream that starts arriving at an endpoint 30+ millseconds ahead of its video stream counterpart(s) will produce detectable lip-synchronization problems for most participants. An audio stream that arrives later than its associated video stream data has a slightly higher tolerance of 40+ milliseconds before the loss of audio and video synchronization becomes generally detectable.
Network Jitter This term is associated with the loss or desequencing of data packets in a real-time data stream. A packet loss rate of 1% produces roughly on Fast Video Update per second for a video stream. This would produce jerky video. Lost audio packets are not recovered and would therefore produce some form of audio loss. Since audio operates with smaller packets at a lower bandwidth, in general, it is usually less likely to encounter network jitter, but an audio stream is not immune from the effects of jitter. A 2% packet jitter rate starts to render the video stream generally unusable, though audio may be minimally acceptable. Any consistent packet loss above 2% is definitely unacceptable for H.323 videoconferencing. Packet loss in the 1-2% should still be considered a poor network environment and the cause of this type of consistent, significant packet loss should be resolved. Network Analysis A network analyzer unit or application and some simple PC-based network tools are all that is necessary for performing the basics of network analysis for H.323. Those readers skilled in utilizing the advanced features of a network analyzer do not necessarily need the advice entailed in this section since the process of network traffic analysis, profiling, and traffic generation is already known to them. The first step is to analyze the network traffic load on a network at various time periods during a day. Using a network analyzer product this can be simply done by using the overall traffic analysis tool, which generates statistics regarding the total number of packets, total number of bytes, network utilization, collisions, runts, etc. Any Ethernet network that exceeds a 30% utilization encounters collisions when using shared media (i.e. hubs). Collisions can contribute to excessive and/or non-uniform latencies for all data traffic. For simple latency determination for round trip estimates, the "ping" utility included with all TCP/IP enabled PCs and Workstations can be used. With "ping" a user can set the packet size and determine transit times to other nodes. Walking the "ping" packet size from 480 bytes to 640 to 880 to 1024 to 1280 to 1500 will generate packet sizes similar to those used in the video and audio data streams. To determine latency by network "hop" (i.e. network segment), a simple method using "tracert", which is another common TCP/IP application, can be used. This tool outputs the number of network segments encountered en route to the target endpoint. Each segment is denoted by the name of the IP Gateway/Router that was involved in the packet forwarding process. This tools gives the latency encountered to reach each segment all the way to the final destination. Running this tool several times along the same path will give the average latency encountered when traversing the interim network segments.