The Quality of Service (QoS) is a crucial subject in the implementation of VoIP. The main concern is how to provide assurance that packet traffic for a media connection such as voice will not experience delays or drops as a result of interference from other traffic that is of lower priority.
The things to be put into consideration include:
Latency: delay for the delivery of packets
Jitter: variations in the delay of delivery of packets
Burstiness of and Jitter: Losses and discards (as a result of jitter) which have the tendency of occurring in bursts.
Packet Loss: excessive traffic in the network which leads to dropping of packets by the network
For end users, huge delays are troublesome and can lead to bad echoes. It is quite difficult to have a plausible conversation whenever the delays are too large. You will keep on interrupting each other in the conversation. Jitter leads to unusual sound effects but with the use of ‘jitter buffers’ in the software, it can somehow be handled. Packet loss leads to interruptions and even so, a certain amount of it may not be noticeable. However, too much of it makes sound lousy.
Requirements of VoIP QoS
Usually, callers start noticing roundtrip voice delays starting from 250ms or higher. The ITU-TG 114 recommends a one-way latency of a maximum of 150ms. Since this also takes account of the whole voice path, part of which could be on the public internet, the transit latencies of your own network should be significantly less than 150ms.
Most network SLAs usually specify maximum latency
Axiowave SLA 65ms max latency
Verio SLA 55ms max latency
Qwest SLA 50ms max latency (40.86ms; Actual Measured for October 2004)
Internap SLA 45ms max latency
The above SLA numbers are for backbone service providers, a VoIP call’s overall latency could also include the extra latency in the local ISP networks of both the VoIP provider and the user.
There are several ways of measuring jitter. Measurement calculations of jitter are defined in:
IETF RFC 3550 RTP – A Transport Protocol for Real-Time Applications
Nonetheless, network and equipment vendors do not often detail precisely how they calculate the values reported for the measured jitter. The majority of VoIP endpoint devices such as ATAs and VoIP phones have jitter buffers for compensation of network jitter.
From Cisco; Jitter buffers (used for compensating for delay variations) further increase the end-to-end delays, and are often only effective on varying delays of lower than 100ms. Therefore, jitter has to be minimized.
What is the acceptable jitter level in a network? Some network providers currently stipulate the maximum level of jitter in their SLAs.
Qwest SLA 2ms max jitter (0.10ms; Actual Measured for October 2004)
Viterla SLA 1ms max jitter
Internap SLA 0.5ms max jitter
Axiowave SLA 0.5ms max jitter
Verio SLA 0.5ms average, 10ms max jitter shouldn’t be exceeded more than 0.1% of the time
The above SLA numbers are for backbone service providers, a VoIP call’s overall jitter could also include the extra jitter in the local ISP networks of both the VoIP provider and the user.
For all-inclusive information on jitter:
Voice Over IP is packet loss intolerant, i.e. even a packet loss of 1% can substantially downgrade the quality of a VoIP call that is using a compressing codec such as the G.711 codec can tolerate even a lesser amount of packet loss.
Cisco says that the default G. 729 codec needs a packet loss of far lower than 1% to evade noticeable errors. If at all possible, there should be no packet loss at all for VoIP.
Typically, the maximum packet loss is specified by the majority of network SLAs:
Qwest SLA 0.5% maximum packet loss (0.03%; Measured Actual for October 2004)
Internap SLA 0.3% maximum packet loss
Verio SLA 0.1% maximum packet loss
Axiowave SLA 0% maximum packet loss
The above SLA numbers are for backbone service providers, a VoIP call’s overall packet loss could also include the extra packet loss in the local ISP networks of both the VoIP provider and the user.
For a deeper look at the packet loss’s fluctuating nature, click here.
The number of solutions is as many as that of network engineers (too many!)
MTS Multiprotocol Test Suite: a testing tool for protocols such as SIP, QoS, Load, Functional and Automation
Hosted VoIP QoS Solution: monitoring occurs from a 24-7-365 NOC
NetEqualizer: a plug-and-play appliance used for detection of congested bandwidth and reprioritizes the traffic to guarantee QoS of VoIP
StarTrinity SIP Tester: a free RTP and SIP monitoring tool with email reports and alerts, RFC3550 global max jitter, G. 107 MOS/R factor, real-time reports and charts.
WebCDR Watchdog: the internet based QoS monitoring for TDM and VoIP switches
Zynknet WidthGuard: Box or Software appliance used to ensure QoS of VoIP as well as all other Real Time Protocols
SoliCall: PBX mate software for improving and monitoring QoS. Compatible with any IP PBX
Xelor Software: automates the deployment, configuration and management of Quality of Service for instantaneous enterprise network communications
Prioritization: the slowest link is usually the first outgoing link. If you send out this link with top priority when you get voice out, the remainder of the hops is usually not problematic
MyVoIPSpeed: internet based connection testing between end users and your server, receive reports of connection quality, packet loss, the number that support VoIP lines, jitters and much more
VoIP Spear: an online service that monitors the quality of your VoIP 24-7-365. It is usage is free for personal needs and quite economical if you want to use it commercially. http://www.voipspear.com
Real Call Quality (Recqual): a call quality tool based on Asterisk. It carries out an analysis of the round trip audio path.
Dynamic QoS (DQoS): automatically detects and allocates bandwidth for VoIP or any other applications working in real-time
AQuA Powered Asterisk VQM (Voice Quality Monitoring) Solution: V/A Difference, R-Value, PESQ, MOS, MySQL for call records, Graph monitoring stats, Open source code, Schedule logic, web interface, Asterisk powered dialer
Network Traffic Tuning: boxes that can be added to a network for management of bandwidth usage to create Quality of Services even if other network devices don’t support it
Addressing QoS Beyond the Provider Network: Patton’s revolutionary QoS technology reduces jitter and delay for delivery of toll-quality voice on each and every call. For more information, read the Patton white paper here and find out why advanced QoS on the Internet access link is vital for higher quality of voice in the VoIP networks
Passive Voice Quality Analyzer (PVQA): it is based on analysis of pure waveform. This software executes a ground-breaking approach for the reception of real-time MOS scores of voice records and provides data on the impairments of voice quality that led to the QoE loss. The Passive Voice Quality Analyzer is a viable substitute for P .563 ITU-T recommendation and more.
Resource Reservation: making sure that the bandwidth needed for the VoIP call is allocated from one point to another to enable the conversation to take place. However, this may only work on a private network, it won’t work on an internet network where there are a lot of providers between end points or service providers who have no contract with either the caller or the recipient of the call.
VoIPMonitor: an open source packet sniffer of live networks which carries out an analysis of RTP and SIP protocol;
Saves every call as a standalone pcap file
Stores detailed statistics of MOS/loss/delay to MySQL
Calculates MOS-LQE score with respect to the ITU-T G .107 E model