QoS (Quality of Service) is a major issue in VOIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interference from other lower priority traffic.

Things to consider are
  • Latency: Delay for packet delivery
  • Jitter: Variations in delay of packet delivery
  • Packet loss: Too much traffic in the network causes the network to drop packets
  • Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in bursts

For the end user, large delays are burdensome and can cause bad echos. It's hard to have a working conversation with too large delays. You keep interrupting each other. Jitter causes strange sound effects, but can be handled to some degree with "jitter buffers" in the software. Packet loss causes interrupts. Some degree of packet loss won't be noticeable, but lots of packet loss will make sound lousy.

VOIP QoS Requirements


Callers usually notice roundtrip voice delays of 250ms or more. ITU-T G.114 recommends a maximum of a 150 ms one-way latency. Since this includes the entire voice path, part of which may be on the public Internet, your own network should have transit latencies of considerably less than 150 ms.

Most network SLAs specify maxium latency
  • Axiowave SLA 65ms maximum latency
  • Internap SLA 45ms maximum latency
  • Qwest SLA 50ms maximum latency - Measured Actual for Oct 2004: 40.86ms
  • Verio SLA 55ms maximum latency
The SLA numbers above are for backbone providers, the total latency for a VOIP call may also include additional latency in the VOIP provider's and the user's local ISP networks.


Jitter can be measured in several ways. There are jitter measurement calculations defined in:
  • IETF RFC 3550 RTP: A Transport Protocol for Real-Time Applications
  • IETF RFC 3611 RTP Control Protocol Extended Reports (RTCP XR)
But, equipment and network vendors often don't detail exactly how they are calculating the values they report for measured jitter. Most VOIP endpoint devices (e.g. VOIP phones and ATAs) have jitter buffers to compensate for network jitter. Quoting from Cisco:
  • Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and are usually only effective on delay variations less than 100 ms. Jitter must therefore be minimized.

Whats an acceptable level of jitter in a network? Several network providers now speciify maximum jitter in their SLAs.
  • Axiowave SLA 0.5ms maximum jitter
  • Internap SLA 0.5ms maximum jitter
  • Qwest SLA 2ms maximum jitter - Measured Actual for Oct 2004: 0.10ms
  • Verio SLA 0.5ms average, not to exceed 10ms maximum jitter more than 0.1% of time
  • Viterla SLA 1ms maximum jitter
The SLA numbers above are for backbone providers, the total jitter for a VOIP call may also include additional jitter in the VOIP provider's and the user's local ISP networks.

Detailed jitter reading

Packet Loss

VOIP is not tolerant of packet loss. Even 1% packet loss can "significantly degrade" a VOIP call using a G.711 codec and other more compressing codecs can tolerate even less packet loss.
Cisco says:
  • The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors. Ideally, there should be no packet loss for VoIP

This link discusses the time varying nature of packet loss http://www.voiptroubleshooter.com/indepth/burstloss.html

Most network SLAs specify maxium packet loss
  • Axiowave SLA 0% maximum packet loss
  • Internap SLA 0.3% maximum packet loss
  • Qwest SLA 0.5% maximum packet loss - Measured Actual for Oct 2004: 0.03%
  • Verio SLA 0.1% maximum packet loss
The SLA numbers above are for backbone providers, the total packet loss for a VOIP call may also include additional packet loss in the VOIP provider's and the user's local ISP networks.


There are as many solutions as there are network engineers (that is, too many :-) )

  • PVQA - Passive Voice Quality Analyzer based on pure waveform analysis. PVQA software implements innovative approach for receiving MOS scores of real time voice records and deliver information on the voice quality impairments that caused the QoE loss. PVQA is a competitive alternative for P.563 ITU-T recommendation and more.
  • Addressing QoS Beyond the Provider Network Read this Patton white paper to learn why advanced Quality of Service (QoS) on the Internet access link is the key to superior voice quality in VoIP networks. Learn how Patton's industry-leading QoS technology minimizes delay and jitter to deliver toll-quality voice on every call.
  • AQuA Powered Asterisk Voice Quality Monitoring Solution - Asterisk powered dialer, web interface, Schedule logic, Open source code, Graphing monitoring stats, MySQL for call records, MOS, PESQ, R-Value, V/A Difference.
  • Dynamic QoS: SmartShare's DQoS makes sure to automatically detect and allocate bandwidth for VoIP or other real-time applications
  • Hosted VoIP Qos Solution monitored from a 24/7/365 NOC http://www.rcnpg.com
  • MyVoIPSpeed - Web-based testing of connections between your server and end-users, get reports of jitter, packet loss and connection quality, the number support VoIP lines and more.
  • NetEqualizer - Plug-and-play appliance that detects bandwidth congestion and reprioritizes traffic to ensure VoIP QoS
  • Network Traffic Tuning Boxes you can add to a network to manage bandwidth usage and create QOS even if the other network devices don't support it.
  • Prioritization: The first outbound link is the slowest. If you get voice out this link with top priority, the remaining hops are usually no problem.
  • Recqual - Recqual (Real Call Quality) is an Asterisk based call quality tool that analyzes the round trip audio path.
  • Resource reservation : to make sure that the VoIP call has the bandwidth needed allocated from point to point before the conversation takes place. This may work on a private network, but will not work on the Internet where there are many providers between end points, providers with no contract agreement with the caller or the callee.
  • SoliCall - PBXMate software to improve & monitor QoS. Works with any IP PBX.
  • VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol.
    • Predicts MOS-LQE score according to ITU-T G.107 E-model
    • Detailed delay/loss/MOS statistics stored to MySQL
    • Each call is saved as standalone pcap file
  • VoIP Spear — web service that monitors your VoIP quality 24x7x365. Free for personal use and very affordable for commercial use. http://www.voipspear.com
  • WebCDR Watchdog - Web-based QoS monitoring for VoIP and TDM switches
  • Xelor Software - Software to automate the configuration, deployment, and management of QoS for realtime communications on enterprise networks.
  • Zynknet WidthGuard - Software or Box appliance to ensure de Quality of Service of VoIP and all others Real Time protocols.
  • MTS Multiprotocol Test Suite: SIP and other protocols testing tool; Functional, Load, Automation, QOS.
  • StarTrinity SIP Tester - Freeware SIP and RTP monitoring tool with G.107 MOS/R-factor, RFC3550 global max jitter, realtime charts and reports, email alerts and reports.

General links

QoS Protocols

QoS Monitoring

QoS Howtos

QoS advice

Qos Engineers

  • QoS on Cisco networks
    • Web:http://www.dcomms.co.uk/DataComms Europe Ltd.
    • Telephone: +44(0)
    • Contact: George Adade
    • Email: enquiries@dcomms.co.uk
    • Offer QoS Advice and installation on all things Cisco. Remote and Onsite configurations. Optimal Cisco configurations, MPLS design and trouble shouting.

  • Australia: PureTel
    • Telephone: +61(0) 3 98999413 begin_of_the_skype_highlighting              +61(0) 3 98999413      end_of_the_skype_highlighting
    • Email: info@puretel.com.au
    • Qualified VoIP and Networking Experts. PureTel provides professional service and support for networking infrastructure that needs to run VoIP traffic.

See Also

Created by: jht2, Last modification: Tue 02 of Feb, 2016 (15:04 UTC) by tenarys
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