Aastra Telecom 57I CT, 55I, 53I manual Real-time Transport Protocol RTP Settings

IP Phone Administrator Guide Global SIP Settings

Configuring the IP Phones

Real-time Transport Protocol (RTP) Settings

Real-time Transport Protocol (RTP) is used as the bearer path for voice packets sent over the IP network. Information in the RTP header tells the receiver how to reconstruct the data and describes how the bit streams are packetized (i.e. which codec is in use). Real-time Transport Control Protocol (RTCP) allows endpoints to monitor packet delivery, detect and compensate for any packet loss in the network. Session Initiation Protocol (SIP) and H.323 both use RTP and RTCP for the media stream, with User Datagram Protocol (UDP) as the transport layer encapsulation protocol.

Note: If RFC2833 relay of DTMF tones is configured, it is sent on the same port as the RTP voice packets.

You can set the following parameters for RTP on the IP Phones:

Aastra Web UI Parameters	Configuration File Parameters
RTP Port	sip rtp port
Basic Codecs (G.711 u-Law, G.711 a-Law, G.729)	sip use basic codecs
Force RFC2833 Out-of-Band DTMF	sip out-of-band dtmf
Customized Codec Preference List	sip customized codec
DTMF Method (global and per-line settings)	sip dtmf method (global and per-line settings)
RTP Encryption (global and per-line settings)	sip srtp mode (global and per-line settings)
Silence Suppression	sip silence suppression

4-64 41-001160-00, Release 2.1, Rev 04