Aastra Telecom 9480i Series User Manual

41-001160-03, Rev 00, Releaes 2.4

4-93

Configuring Network and Session Initiation Protocol (SIP) Features

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.

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

If RFC2833 relay of DTMF tones is configured, it is sent on the 

same port as the RTP voice packets. The phones support decoding and 
playing out DTMF tones sent in SIP INFO requests. The following 
DTMF tones are supported:
•

Support signals 0-9, #, *

•

Support durations up to 5 seconds

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

Draft 1