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

Latency

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

• Qwest SLA 50ms maximum latency - Measured Actual for Oct 2004: 40.86ms
• Axiowave SLA 65ms maximum latency
• Verio SLA 55ms maximum latency
• Internap SLA 45ms 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

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.

• Qwest SLA 2ms maximum jitter - Measured Actual for Oct 2004: 0.10ms
• Viterla SLA 1ms maximum jitter
• Axiowave SLA 0.5ms maximum jitter
• Verio SLA 0.5ms average, not to exceed 10ms maximum jitter more than 0.1% of time
• Internap SLA 0.5ms 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

• Nice overview here
• More detailed overview http://www.voiptroubleshooter.com/indepth/jittersources.html

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. (Intel whitepaper)
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 (Cisco Whitepaper)

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

Most network SLAs specify maxium packet loss

• Qwest SLA 0.5% maximum packet loss - Measured Actual for Oct 2004: 0.03%
• Axiowave SLA 0% maximum packet loss
• Verio SLA 0.1% maximum packet loss
• Internap SLA 0.3% 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.

Solutions

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

• 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.
• 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.
• 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.
• Hosted VoIP Qos Solution monitored from a 24/7/365 NOC http://www.rcnpg.com
• Xelor Software - Software to automate the configuration, deployment, and management of QoS for realtime communications on enterprise networks.
• 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.

General links

• Nice overview here

QoS Protocols

• RSVP : Resource Reservation Protocol, IETF
• DiffServ: Differentiated Services IETF
• MPLS: Multiprotocol Label Switching IETF
• SBM
• IP QoS: QoS in IP headers

QoS Monitoring

• See: How To Debug and Troubleshoot VOIP
• Online VoIP voice quality test http://www.testyourvoip.com/
• Hosted VoIP Qos Solution monitored from a 24/7/365 NOC http://www.rcnpg.com
• Embedded VoIP monitoring software http://www.telchemy.com
• MyVoIPSpeed online VoIP connection test: reports jitter, packet loss, bandwidth quality http://myvoipspeed.visualware.com

QoS Howtos

• How to implement QoS to get crystal clear VOIP sound on your DSL / Internet connection.
• Linux Router: Build a powerful low-cost router using linux
• Ztelco Administrators have written a simple easy to use QoS Script. It worked for me. Check it out.

QoS advice

• QoS on Linux systems
  • The Wonder Shaper works wonders for VOIP too
  • QoS Linux: with HTB
  • QoS Linux with HFSC: Simple Script
  • Another attempt at QoS with Linux using PRIO and HTB
• QoS FreeBSD: QoS on FreeBSD
• QoS Cisco: QoS on Cisco Routers
• QoS Cisco IOS: QoS on Cisco IOS (Routers)
• QoS Windows: Bandwidth management and QoS on Linux systems
• QoS WLAN: Quality of Service in Wireless LANs (WiFi)
• QoS Ethernet: Quality of Service in Ethernet networks
• http://www.voiptroubleshooter.com/ VoIP Troubleshooter
• Asterisk QoS: Asterisk QoS information
• Xelor Software - Software to automate the configuration, deployment, and management of QoS for realtime communications on enterprise networks.

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
  • 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.

• Colombia: Quintum, 3com and cisco QoS binnacleita
  • Telephone+57(1)5221223
  • Email:comercial@binnacleita.com

See Also

• VLAN: Larger installations typically use a Virtual LAN for VoIP quality purposes
• VOIP Routers Routers that have QoS support