Monitoring of VoIP Call Quality
Global SIP Statistics
The Allegro Network Multimeter displays all significant quality metrics such as packet loss and jitter globally. In addition, it displays all successful and unsuccessful SIP calls according to error code via a graph and allows SIP errors to be narrowed down to a single point in time. In addition, all simultaneous telephone calls are displayed, e.g. to detect an overload caused by too many simultaneous telephone calls.
All SIP calls are clearly displayed in a table. All calls can be easily sorted and filtered according to call IDs, phone numbers, IP addresses, call IDs and jitter or MOS values. The SIP recognition automatically correlates the RTP connection belonging to the SIP call and displays this graphically with packet rate, packet loss, MOS value and jitter. The SIP and RTP connections can be extracted as a pcap.
Analysis of SIP and RTP Connections
The Allegro Network Multimeter has integrated analysis for both SIP and correlated RTP connections. This allows the exact sequence of server communication to be tracked in real-time, even during individual SIP calls. For example, it is possible to check whether the authentication was carried out on the SIP server and was successful.
Analysis of Poor Speech Qualities
A classic application is a telephone call a few minutes or hours ago which exhibited poor voice quality. The Allegro Network Multimeter can extract this call from millions of others by real-time search and display the packet loss, jitter and MOS value of the RTP connection. If all these values are correct, the audio stream can be examined to see whether the source of the error is not outside the SIP system and, for example, there is already noise, crackling or an insufficient RTP level.
Here you find: