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SIP User's Manual
7. IP Telephony Capabilities
The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide
a good compromise between delay and error rate. The jitter buffer ‘holds’ incoming packets
for 10 msec before making them available for decoding into voice. The coder polls frames
from the buffer at regular intervals in order to produce continuous speech. As long as
delays in the network do not change (jitter) by more than 10 msec from one packet to the
next, there is always a sample in the buffer for the coder to use. If there is more than 10
msec of delay at any time during the call, the packet arrives too late. The coder tries to
access a frame and is not able to find one. The coder must produce a voice sample even if
a frame is not available. It therefore compensates for the missing packet by adding a BadFrame-Interpolation (BFI) packet. This loss is then flagged as the buffer being too small.
The dynamic algorithm then causes the size of the buffer to increase for the next voice
session. The size of the buffer may decrease again if the device notices that the buffer is
not filling up as much as expected. At no time does the buffer decrease to less than the
minimum size configured by the Minimum delay parameter.
For certain scenarios, the Optimization Factor is set to 13: One of the purposes of the
Jitter Buffer mechanism is to compensate for clock drift. If the two sides of the VoIP call are
not synchronized to the same clock source, one RTP source generates packets at a lower
rate, causing under-runs at the remote Jitter Buffer. In normal operation (optimization factor
0 to 12), the Jitter Buffer mechanism detects and compensates for the clock drift by
occasionally dropping a voice packet or by adding a BFI packet.
Fax and modem devices are sensitive to small packet losses or to added BFI packets.
Therefore, to achieve better performance during modem and fax calls, the Optimization
Factor should be set to 13. In this special mode the clock drift correction is performed less
frequently - only when the Jitter Buffer is completely empty or completely full. When such
condition occurs, the correction is performed by dropping several voice packets
simultaneously or by adding several BFI packets simultaneously, so that the Jitter Buffer
returns to its normal condition.
7.7
Configuring Alternative Routing (Based on
Connectivity and QoS)
The Alternative Routing feature enables reliable routing of Tel-to-IP calls when a Proxy isn’t
used. The device periodically checks the availability of connectivity and suitable Quality of
Service (QoS) before routing. If the expected quality cannot be achieved, an alternative IP
route for the prefix (phone number) is selected.
7.7.1
Alternative Routing Mechanism
When a Tel-to-IP call is routed through the device, the call’s destination number is
compared to the list of prefixes defined in the 'Tel to IP Routing' table (described in ''Tel to
IP Routing Table'' on page 156). The 'Tel to IP Routing' table is scanned for the destination
number’s prefix starting at the top of the table. For this reason, enter the main IP route
above any alternative route. When an appropriate entry (destination number matches one
of the prefixes) is found, the prefix’s corresponding destination IP address is verified. If the
destination IP address is disallowed (or if the original call fails and the device has made
two additional attempts to establish the call without success), an alternative route is
searched in the table. , after which an alternative route is used.
Destination IP address is disallowed if no ping to the destination is available (ping is
continuously initiated every seven seconds), when an inappropriate level of QoS was
detected, or when a DNS host name is not resolved. The QoS level is calculated according
to delay or packet loss of previously ended calls. If no call statistics are received for two
minutes, the QoS information is reset.
Version 5.4
331
May 2008