New feature release: Quality of Service (QoS) Reports for Voice Calls

At Telnyx, we know that it’s important for our customers to be able to track all aspects of their calls. Through our partnership with QXIP, we've developed Quality of Service (QoS) reports for voice calls to help users troubleshoot call quality issues that may arise.

Following on from the Winter release of the Advanced SIP Debugging Tool, this new feature will give customers a visual representation of the Quality of Service (QoS) of any given call on a timeline with two separate graphics, one for each the RTP media streams that are established for each call.

The new report feature is available in the Mission Control Portal through the Advanced View of our SIP Debugging tool. Reports are based on RTCP reports that are sent between SIP devices and Telnyx:

RTP streams and RCTP reports between Telnyx and the user

When a call is established media starts flowing over two RTP streams, one in each direction. Periodically, each side also sends RTCP reports to report transmission and reception statistics during the interval. Each report includes the number of packets sent (packets), the accumulated packets that were expected but never received (packets_lost), the jitter perceived from the received packets (ia_jitter), and more.

The QoS report feature uses the data on these RTCP reports to display 4 different quality metrics on a timeline for each RTP stream. Let’s take a look at what these metrics mean:


Mean Opinion Score (MOS) is a value that represents the audio quality of the call as perceived by the reporter. it's calculated based on Jitter and Packets Lost values. The scale goes from 1 (Bad) to 4.5 (Excellent).

Mean Opinion Score

IA Jitter

“Ia_jitter” represents the value of Interarrival Jitter, as measured by the reporter.

IA Jitter

Packets Lost

“Packets Lost” represents the accumulated number of RTP packets perceived as lost by the reporter.

Packets Lost


“Packets” represents the accumulated number of RTP packets that were sent by the reporter.


These quality metrics will enable users to quickly identify any issues that might have affected the quality of the call, which will speed up the troubleshooting process.

Please note that in order to capture RTCP data from your calls you first need to enable RTCP capture for your Connections on the Mission Control Portal. You can do this by Editing your SIP connection and turning the RTCP Capture toggle to ‘on’.

Edit your SIP connection. Turn the RTCP Capture toggle to 'on'.

If you have any questions please don’t hesitate to reach out to our team of experts.

QXIP (QuickSIP) is an R&D company that specializes in open-source and commercial development of voice monitoring technology, based in Amsterdam, The Netherlands.