Global SIP Settings

RTCP Summary Reports

The IP phones include the capability of enabling/disabling the generation of RTCP summary reports using the SIP vq- rtcpxr event package. These RTCP summary reports include voice quality statistics according to draft-ietf-sipping-rtcp- summary-05 specifications including packet loss, jitter, and delay statistics, as well as call quality scores. When this fea- ture is enabled, an RTCP summary report is sent at the end of each call via a PUBLISH message to the configuration server.

In addition to enabling/disabling the generation of these reports, you must specify the hostname and port of the entity, known as the collector, receiving the reports. Similar to the other IP Phone SIP account parameters, the RTCP summary report parameters can be set on a global or a per-line basis using the configuration files only.

The RTCP summary report parameters are:

sip rtcp summary reports

sip LineN rtcp summary reports

sip rtcp summary report collector

sip LineN rtcp summary report collector

sip rtcp summary report collector port

sip LineN rtcp summary report collector port

Note:

The transport protocol used for RTCP summary reports is also configurable. Refer to Configurable Transport Protocol for SIP Services and RTCP Summary Reports on page 6-40for more information.

Limitations

The following is a limitation when enabling RTCP summary reports on the phone:

The call must be at least 5 seconds long in order to generate the RTCP extended reports.

Configuring RTCP Summary Reports

Use the following procedure to configure RTCP summary reports.

Configuration Files

For specific parameters you can set in the configuration files, see Appendix A, the section, “RTCP Summary Reports” on page A-37.

Autodial Settings

The IP phones include a feature called “Autodial”. When you configure Autodial on an IP phone, the phone automatically dials a pre-configured number whenever it is off-hook. Depending on the configuration you specify, the Autodial func- tions as either a “hotline”, or as a “warmline,” as follows:

Hotline (default): The IP phone immediately dials a preconfigured number when you lift the handset.

Warmline: The IP phone waits for a specified amount of time after you lift the handset before dialing a pre-configured number. If you do not dial a number within the time allotted, then the IP phone begins to dial the number.

By default, the Autodial feature functions as a hotline. If you want Autodial to function as a warmline, you can use the Autodial “time-out” parameter to specify the length of time (in seconds) the IP phone waits before dialing a pre-config- ured number.

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Aastra Telecom 41-001343-02 manual Autodial Settings, Limitations, Configuring Rtcp Summary Reports