Real-Time Transport Protocol (Rtp) Settings - Aastra 9143i Series Administrator's Manual

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.
You can set the following parameters for RTP on the IP Phones:
Aastra Web UI Parameters
RTP Port
Basic Codecs (G.711 u-Law, G.711 a-Law, G.729)
Force RFC2833 Out-of-Band DTMF
Customized Codec Preference List
DTMF Method (global and per-line settings)
RTP Encryption (global and per-line settings)
Silence Suppression
41-001160-03 Rev 00, Release 2.5
Configuring Network and Session Initiation Protocol (SIP) Features
Note: 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
Configuration File Parameters
sip rtp port
sip use basic codecs
sip out-of-band dtmf
sip customized codec
sip dtmf method (global and per-line settings)
sip srtp mode (global and per-line settings)
sip silence suppression
Global SIP Settings
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