RTP Streams

Note: The VoIP analysis module is only available to VoIP license users or evaluation version users who selected VoIP evaluation mode.

The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. While protocols like SIP or H.323 are used to control the call (e.g. setting up a connection, dialing, disconnecting, etc.), RTP is used for reliable transmission of data packets and maintaining Quality of Service. In other words, RTP streams carry the actual voice payload encoded utilizing one of a number of codecs, and analysis of RTP data provided invaluable information for assessing call quality and troubleshooting VoIP networks.

To view RTP streams captured by the application, select RTP Streams in the left pane of the VoIP analyzer window:

rtp1

The upper part displays a complete list of all RTP streams. When selecting a RTP stream from the list, the lower pane displays detailed information on the selected stream, including the complete list of RTP packets, summarized and statistical data, as well as the charts:

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Up to four different charts for the selected stream can be displayed simultaneously, with the window interval from 5 to 60 seconds. Note that right clicking and dragging the graph will scroll it to the left or right respectively. The following chart types are available:

Packet Count – number of RTP packets per second including duplicates, lost packets, and "out of order" packets.

Stream Bandwidth – stream speed in terms of kilobits per second.

Packet Sizes – average sizes of RTP packets broken down by network and RTP headers, and RTP payload.

Jitter – stream jitter.

R-Factor, MOS Score – stream quality estimation.

Packet Intervals – temporal allocation of RTP packets in a stream.

The RTP Streams list contains all captured RTP streams, both belonging to SIP or H.232 signaling sessions, and the ones for which signaling sessions were not identified (so called 'orphan' streams, i.e. the ones that don't belong to any parent session). Please refer to the Settings chapter for more detailed information on how to exclude RTP streams that don't have corresponding signaling sessions.