Basic Voice Flow
The flow of a compressed voice circuit is shown in this diagram. The analog signal from the telephone is digitized into pulse code modulation (PCM) signals by the voice coder-decoder (codec). The PCM samples are then passed to the compression algorithm which compresses the voice into a packet format for transmission across the WAN. On the far side of the cloud the exact same functions are performed in reverse order. The entire flow is shown in below

Based on how the network is configured, the router/gateway can perform both the codec and compression functions or only one of them. For example, if an analog voice system is used, then the router/gateway performs the CODEC function and the compression function as shown in figure below

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
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.
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 ,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.
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.
Solutions
There are as many solutions as there are network engineers (that is, too many
)
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.
Downloadable Bandwidth Management Apps from Tenarys protect VoIP calls from competing traffic on your Internet connection better than QoS can. The apps are available as a monthly subscription and can be run on a low cost or repurposed PC (P4 1GB).
Dynamic QoS: SmartShare’s DQoS makes sure to automatically detect and allocate bandwidth for VoIP or other real-time applications
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
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.
QoS Monitoring
AQuA – Audio Quality Analyzer to monitor voice quality in VoIP http://www.sevana.fi/aqua_wiki.php
Embedded VoIP monitoring software http://www.telchemy.com
Hosted VoIP Qos Solution monitored from a 24/7/365 NOC http://www.rcnpg.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
Source : http://voip.info.org, http://www.cisco.com