Administrator's Guide

Chapter 10. SIP Services
Allowed number of media streams per SIP session
Enter the number of media streams a single SIP session can handle. This restriction is pri-
marily made for preventing DOS attacks.
Timeout for one-way media streams
This setting is used by the Telecommuting Module to detect when media is only sent in one
direction. If no media packets are received in one direction during the configured number of
seconds, the Telecommuting Module creates a log message about this.
Tear down media streams at RTP/RTCP timeout
Here, you select if the Telecommuting Module should tear down media streams when the
Timeout for RTP streams and Timeout for RTCP streams have been reached.
When the media streams are torn down, the session is still not terminated by the Telecom-
muting Module. This means that there will be no SIP messages sent out (like a BYE) to
indicate that the streams were torn down.
Timeout for RTP streams
This setting is used by the Telecommuting Module to detect a closed media session, even
when no signaling for this was made. If no RTP packets are received during the configured
number of seconds, the Telecommuting Module creates a log message about this. If Tear
down media streams at timeout was selected, the Telecommuting Module will also tear
down the session when the RTP and RTCP timeouts have been reached.
Timeout for RTCP streams
This setting is used by the Telecommuting Module to detect a closed media session, even
when no signaling for this was made. If no RTCP packets are received during the configured
number of seconds, the Telecommuting Module creates a log message about this. If Tear
down media streams at timeout was selected, the Telecommuting Module will also tear
down the session when the RTP and RTCP timeouts have been reached.
Limitation of RTP Codecs
You might want to limit the use of some media codecs. There can be several reasons for
this: some endpoints do not support the codecs, too many codec offers make the SIP request
packet too large (which causes it to be fragmented), they consume too much bandwidth, or
you want to allow only codecs with good enough voice quality.
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