Novell LINUX ENTERPRISE DESKTOP 10 SP1 - GNOME 23-05-2007, Manual

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166 SUSE Linux Enterprise Desktop 10 SP1 GNOME User Guide
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The voice output on both ends sounds strangely clipped.
Try to adjust the jitter buffer using RTP properties in Preferences > Network to compensate for
delayed voice packages. When doing this, be aware that it increases the latency.
DTMF does not work.
You tried to check your voice mail using the DTMF pad, but the connection could not be
established. There are three different protocols used for the transmission of DTMF data, but
only two of these are supported by Linphone (SIP INFO and RTP rfc2833). Check with your
provider whether it supports one of these. The default protocol used by Linphone is rfc2833,
but if that fails you can set the protocol to SIP INFO in
Preferences > Network > Other . If it
does not work with either of them, DTMF transmission cannot be done using Linphone.
7.2.7 Glossary
Find some brief explanation of the most important technical terms and protocols mentioned in this
document:
VoIP
VoIP stands for voice over Internet protocol . This technology allows the transmission of
ordinary telephone calls over the Internet using packet-linked routes. The voice information is
sent in discrete packets like any other data transmitted over the Internet via IP.
SIP
SIP stands for session initiation protocol . This protocol is used to establish media sessions over
networks. In a Linphone context, SIP is the magic that triggers the ring at your counterpart's
machine, starts the call, and also terminates it as soon as one of the partners decides to hang up.
The actual transmission of voice data is handled by RTP.
RTP
RTP stands for real-time transport protocol . It allows the transport of media streams over
networks and works over UDP. The data is transmitted by means of discrete packets that are
numbered and carry a time stamp to allow correct sequencing and the detection of lost
packages.
DTMF
A DTMF encoder, like a regular telephone, uses pairs of tones to represent the various keys.
Each key is associated with a unique combination of one high and one low tone. A decoder
then translates these touch-tone combinations back into numbers. Linphone supports DTMF
signalling to trigger remote actions, such as checking voice mail.
codec
Codecs are algorithms specially designed to compress audio and video data.
jitter
Jitter is the variance of latency (delay) in a connection. Audio devices or connection-oriented
systems, like ISDN or PSTN, need a continuous stream of data. To compensate for this, VoIP
terminals and gateways implement a jitter buffer that collect the packets before relaying them
onto their audio devices or connection-oriented lines (like ISDN). Increasing the size of the
jitter buffer decreases the likelihood of data being missed, but the latency of the connection is
increased.