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In this above form that (the one you get by the clicking on RTP main menu entry), you can set up the voice transport features.
Rtpstart and RtpEnd
Rtpstart and Rtpend define the RTP port range reserved for RTP traffic.
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18000 - 15000 = 3000 ports available. This means 3000/4 = 750 concurrent calls threshold.
Rtpchecksums
The Rtpchecksums enable the application checksum over UDP encrypted voice transmission. This is an error detection commodity, which adds 16 bits per packets payload.
Strictrtp
Strictrtp Enables the strict RTP protection. This will drop RTP packets that do not come from the source of the RTP stream. This option is disabled by default.
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In order to enable call roaming through different kind of networks (mobiles 'X'G family and Wi-Fi) you need to DISABLE Strictrtp. |
Strictrtp option is about call roaming. If it's enabled then each call would be terminated on network change event.
Rtp timeout
Rtp timeout sets up the amount of seconds one server can wait without dealing RTP packages on one call's leg. This value is connected to client's setup and it gives you the fault tolerance time of one call.
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Rtp timeout is used in combination with Strictrtp to enable call roaming: during network change events RTP stream is lost until one new UDP connection is established between client and server. Rtp timeout is the maximum amount of seconds one server can wait before considering such connection as lost and thus hang up any related ongoing call. |
2.4.3 Jitter Buffer
Jitter is the variation in latency as measured in the variability over time of the packet latency across a network. The consequences of jitter, often called jittering, are a voice communication with gaps in it or stirring metal voice effect. Mostly on a GPRS and EDGE network (and in general in a mobile network environment), the jitter is a sensible problem to face. To cope with jittering issues, a jitter buffer produces a smooth and regular audio output, just adding some more latency.
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