Advanced Network Settings4-50 41-001129-00 Rev 09, Release 1.4.1IP Phone Administrator GuideConfiguring the IP Phones Real-time Transport Protocol (RTP) SettingsReal-time Transport Protocol (RTP) is used as the bearer path for voice packetssent over the IP network. Information in the RTP header tells the receiver how toreconstruct the data and describes how the bit streams are packetized (i.e. whichcodec is in use). Real-time Transport Control Protocol (RTCP) allows endpointsto monitor packet delivery, detect and compensate for any packet loss in thenetwork. Session Initiation Protocol (SIP) and H.323 both use RTP and RTCP forthe media stream, with User Datagram Protocol (UDP) as the transport layerencapsulation protocol.RTP PortRTP is described in RFC1889. The UDP port used for RTP streams is traditionallyan even-numbered port, and the RTCP control is on the next port up. A phone calltherefore uses one pair of ports for each media stream.On the Aastra IP phone, the initial port used as the starting point for RTP/RTCPport allocation can be configured using "RTP Port Base". The default RTP baseport on the IP phones is 3000.For example, if the RTP base port value is 5000, the first voice patch sends RTPon port 5000 and RTCP on port 5001. Additional calls would then use ports 5002,5003, etc.You can configure the RTP port on a global-basis only, using the configurationfiles, the IP Phone UI, or the Aastra Web UI.Basic CodecsCODEC is an acronym for COmpress-DECompress. It consists of a set ofinstructions that together implement one or more algorithms. In the case of IPtelephony, these algorithms are used to compress the sampled speech data, todecrease the content's file size and bit-rate (the amount of network bandwidth inkilobits per second) required to transfer the audio. With smaller file sizes andlower bit rates, the network equipment can store and stream digital media contentover a network more easily.Note: If RFC2833 relay of DTMF tones is configured, it is sent on thesame port as the RTP voice packets.
