Inalp Networks SmartWare Release 2.10, Руководство по настройке программного обеспечения

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Gateway Configuration 287
accommodate different target compression rates such as 8, 6.4 and even 5.3 kbps. Note that
this bit rate is for audio only. When transmitting packetized voice over the network, protocol
overhead (such as RTP/UDP/IP/Ethernet) is added on top of this bit rate, resulting in a
higher actual data rate.
• Complexity – the higher the complexity of implementing the codec, the more CPU resources
are required.
• Voice quality – compressing voice in some codecs results in very good voice quality, while
others cause a significant degradation.
•
Digitizing delay – Each algorithm requires that different amounts of speech are buffered
prior to the compression. This delay adds to the overall end-to-end delay. A network with
excessive end-to-end delay, often causes people to revert to a half-duplex conversation
(“How are you today? over…”) instead of the normal full-duplex phone call.
Which codec you can use depends on the VoIP gateway. In ISoIP there are more codecs available
than in H.323. Codecs can be set in the VoIP gateways as well as in the CS interfaces.
When the codec is configured, you may also set the packet size. In SmartWare it is possible to specify
the packet size of the transmitted voice packets. Larger packet sizes significantly reduce the overall
bandwidth but add to the packetization delay as the sender needs to wait longer to fill up the
payload. In contrast to broadcast-type media transmission (e.g. RealAudio), a two-way phone
conversation is sensitive to latency.
Most callers notice round-trip delays when they exceed 250 ms, so the one-way latency budget
would typically be 150 ms. Beyond that round-trip latency, callers start feeling uneasy when holding
a two-way conversation and usually end up talking over each other. At 500 ms round-trip delays and
beyond, phone calls are impractical. For reference, the typical delay when speaking through a geo-
stationary satellite is 150-500 ms. For this reason be sure you understand the effects of changing the
packet size before you change it. The default value is usually suitable so that you don't have to
specify the packet length. For more information on default values refer to Chapter 20, “Gateway
ISoIP Mode”, and Chapter 21, “Gateway H.323 Mode” in the SmartWare
Command Reference Guide .
Note: There is no right CODEC. The choice of what compression scheme to use depends on what
parameters are more important for a specific installation. In practice, G.723 and G.729 are more
popular than G.726 and G.728. For an overview of used codecs in SmartWare see Chapter 36.2,
“Available Voice Codecs in SmartWare 2.” later in this guide.
30.3.2 Configure used Codec for an ISoIP Connection
The codec used for an ISoIP connection could be set in the ISoIP gateway as well as in the CS
interface as follows. In the ISoIP gateway a default codec is set. If you do not specify a codec in the
ISoIP interface, the default codec specified in the ISoIP gateway will be used. Otherwise the codec
specified in the ISoIP gateway will be replaced by the codec specified in the ISoIP interface. In the
same way it is possible to specify the packet size of the transmitted voice packets.
Procedure
To configure the used codec for an ISoIP connection
Mode
Gateway ISoIP
Command Purpose
Step 1 node (gw-isoip)[isoip]#codec <?>
List all available codecs on the ISoIP gateway
Software Configuration Guide Release 2.10, Revision 1.00