QoS - Quality of Service - in VOIP implementations.

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 to 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 maximum 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

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.

What’s an acceptable level of jitter in a network? Several network providers now specify 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

More detailed overview http://www.voiptroubleshooter.com/indepth/jittersources.html
RTP and Jitter explained https://kb.smartvox.co.uk/voip-sip/rtp-jitter-audio-quality-voip/

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 maximum 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.

Solutions

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 an 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.
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
NetEqualizer – Plug-and-play appliance that detects bandwidth congestion and reprioritizes traffic to ensure VoIP QoS
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.
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.
Obkio VoIP Monitoring is a solution from Obkio’s Network Performance Monitoring software which measures VoIP Quality using MOS Score and monitors QoS for each Class of Service using DSCP tags.

General links

Nice overview here

QoS Protocols

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

QoS Monitoring

Embedded VoIP monitoring software http://www.telchemy.com
How To Debug and Troubleshoot VOIP
MyVoIPSpeed online VoIP connection test: reports jitter, packet loss, bandwidth quality http://myvoipspeed.visualware.com
SoliCall Monitors QoS http://www.solicall.com
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

QoS Howtos

QoS with Linux using PRIO and HTB
How to implement QoS to get crystal clear VOIP sound on your DSL / Internet connection. (dead link)
Linux Router: Build a powerful low-cost router using linux. (outdated)
Ztelco Administrators have written a simple easy to use QoS Script. It worked for me. Check it out. (dead link)
Implement QoS for 3CX VoIP on Debian Linux
Implement QoS for 3CX VoIP on Windows

QoS advice

Asterisk QoS: Asterisk QoS information
QoS Cisco: QoS on Cisco Routers
QoS Cisco IOS: QoS on Cisco IOS (Routers)
QoS Ethernet: Quality of Service in Ethernet networks
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 WLAN: Quality of Service in Wireless LANs (WiFi)
VoIP Troubleshooter
Xelor Software – Software to automate the configuration, deployment, and management of QoS for realtime communications on enterprise networks.

Qos Engineers

Australia: PureTel
- Telephone: +61(0) 3 98999413 begin_of_the_skype_highlighting +61(0) 3 98999413 end_of_the_skype_highlighting
- Email: [email protected]
- 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:[email protected]