When girlfriend walk into the VoIP world, you conference a whole brand-new host of protocol standards. Think that the Real-time carry Protocol (RTP) and Real-time Transport control Protocol (RTCP) as the protocols of voice. RTP operates in ~ the deliver layer that the OSI version on height of UDP. Having two deliver layer protocols is a bit odd, but that"s exactly what is happening here. UDP offers the solutions it always does: port numbers (that is, conference multiplexing) and also header checksums (which ensure the the header information does not come to be corrupted). RTP to add time stamps and also sequence number to the header information. This permits the remote an equipment to put the packets ago in order when it obtain them at the remote end (function the the sequence number) and also use a buffer to remove jitter (slight delays) between the packets to give a smooth audio playout (function of the moment stamp). Figure 7.4 provides a visual depiction of the RTP header information contained in a packet.
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The Payload form field in the RTP header is supplied to clues what kind of RTP is in use. You can use RTP for audio or video purposes.
Once two tools attempt to develop an audio session, RTP engages and chooses a random, also UDP harbor number from 16,384 to 32,767 for each RTP stream. Store in mind that RTP streams are one-way. If friend are having actually a two-way conversation, the tools will establish twin RTP streams, one in each direction. The audio stream remains on the at first chosen port for the expression of the audio conference (the devices will not dynamically adjust ports throughout a phone call).
At the moment the devices create the call, RTCP also engages. When this protocol sounds very important, its major job is statistics reporting. That delivers statistics in between the two gadgets participating in the call, i m sorry include
■ Packet count
■ Packet delay
■ Packet loss
■ Jitter (delay variations)
Although this info is useful, that is not almost as vital as the actual RTP audio streams. Keep this in mind as soon as you room configuring QoS settings.
As the devices develop the call, the RTP audio streams usage an even UDP harbor from 16,384 to 32,767, together previously disputed in this section. RTCP create a different session over UDP in between the two tools by making use of an odd-numbered harbor from the same range. Transparent the contact duration, the gadgets send RTCP packets at least once every 5 seconds. The CME router can log and report this information, which enables you to recognize the concerns that room causing contact problems (such as negative audio, call disconnects, and so on) on the network.
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Note RTCP will usage the odd-numbered port adhering to the RTP port. For example, if the RTP audio supplies port 17,654, the RTCP port for the session will certainly be 17,655.