Real-Time Transport Protocol (Rtp) Settings - Mitel 6800 Series Administrator's Manual

Mitel 6800 Series SIP Phone Release 4.2.0 SP2 Administrator Guide
For information about setting this feature, see Chapter 5, the section,
on
page 5-306
25. Click Save Settings to save your changes.

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. 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
You can set the following parameters for RTP on the IP Phones:
MITEL WEB UI PARAMETERS
RTP Port
Basic Codecs (G.711 u-Law, G.711 a-Law,
G.729)
AMR and AMR-WB (G.722.2) Codecs (Licensed
feature)
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
RTP PORT
RTP is described in RFC1889. The UDP port used for RTP streams is traditionally an
even-numbered port, and the RTCP control is on the next port up. A phone call therefore uses
one pair of ports for each media stream.
The RTP port is assigned to the first line on the phone, and is then incremented for each
subsequent line available within the phone to provided each line a unique RTP port for its own
use.
On the IP phone, the initial port used as the starting point for RTP/RTCP port allocation can be
configured using "RTP Port Base". The default RTP base port on the IP phones is 3000.
4-85
.
CONFIGURATION FILE PARAMETERS
sip rtp port
sip use basic codecs
sip amr codec payload format
sip amr codec mode set
sip amr wb codec mode set
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
"XML SIP Notify Events"