In this showdown we take a look at QoS as it relates to VoIP is study some of the competing ISP (Internet Service Provider) vendors.

In the fields of packet switched networks term Quality of Service, abbreviated QoS, refers to resource reservation control mechanisms rather than the achieved service quality. Quality of Service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. Quality of Service guarantees are important if the network capacity is limited, for example in cellular data communication, especially for real-time applications, such as VoIP, since these often require fixed bit rate and are delay sensitive.

A network or protocol that supports Quality of Service may agree on a traffic contract with the application software and reserve capacity in the network nodes, for example during a session establishment phase. During the session it may monitor the achieved level of performance, for example the data rate and delay, and dynamically control scheduling priorities in the network nodes. It may release the reserved capacity during a tear down phase.

In the field of telephony, QoS was defined in the ITU standard X.902 as "A set of quality requirements on the collective behavior of one or more objects". Quality of Service comprises all the aspects of a connection, such as time to provide service, voice quality, echo, loss, reliability and so on. A subset of telephony QoS is Grade of Server(GOS), which comprises aspects of a connection relating to the capacity of a network.

The term Quality of Service is sometimes used as a quality measure, with many alternative definitions, rather than referring to the ability to reserve resources. Quality of Service sometimes refers to the level of Quality of service, i.e. the guaranteed service quality. High QoS is often confused with a high level of performance or achieved service quality.

• 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

In the ISP space, the concept of SLA (Service Level Agreement) is common. This term is a contract bound to certain guaranteed service levels to the end service consumer; QoS plays a role in this domain.

In the VoIP space, 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. Let's take a look at fundamental VoIP QoS requirements and how competing vendors compare in terms of SLA offerings.

VoIP Qos Requirements

Latency is the time taken for a packet to traverse a network from one destination to another. For VoIP, 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. The ISP vendors we're looking at have SLA's that specify maximum latency as follows:

• 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.

Winner: Qwest

Jitter is an unwanted variation of one or more siganl characteristics in telecommunications. Jitter may be seen in characteristics such as the interval between successive pulses, or the amplitude, frequency, or phase of successive cycles. Bottom line is that is sucks when you get jiiter, cause with jitter comes echo, and then you start talking to  yourslef on the phone. Not so fun. 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. phones an ATA's) have jitter buffers to compensate for network jitter. So what is an acceptable level of jitter in a network? The ISP vendors we're looking at have SLA's that specify maximum jitter as follows:

• 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.

 Winner: Qwest

Packet Loss occurs when one or more packets of data traveling across the network fail to reach their destination. 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. 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. The ISP vendors we're looking at have SLA's that specify maximum packet loss as follows:

• 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.

Winner: Qwest

There are as many solutions to the QoS problem depending on where you actually sit in the whole picture. If, for example, you are the service provider then you are most likely implementing as many sophisticated strategies as possible to enusre that your end customer's SLA(s) are met. However, if you are an end consumer or an enterpise that is simply trying to maximize VoIP network effiency there are few suggestions I can make but won't go into detail (I hate to rehash what has already be hashed out). Combine google and an afternoon and study the following terms:

• 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.
• Packet prioritization: Here, the end point suggest a priority on the packets and each router decides if it will honour this request or not.
• 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
• RSVP: Resource Reservation Protocol
• DiffServ: Differentiated Services
• MPLS: Multiprotocol Label Switching
• SBM: Subnet Bandwidth Manager
• IP QoS: QoS in IP headers

Now that I've gone on a tangent let's get back to a conclusion of the showdown. Which ISP vendor has the best SLA? Well on paper it's clear.

Overall Winner: Qwest

Feel free to This e-mail address is being protected from spam bots, you need JavaScript enabled to view it and stay tuned for more BleedingVoIP Showdowns.

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