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 primarily 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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