Nick Goodenough, Spitfire Partner Service Director, explains how to ensure ISDN-level voice quality when deploying a cloud-based VoIP solution
The Cloud offers a seemingly ideal environment for deploying a voice solution but how do we ensure that the user experience is of consistently high quality?
Deploying a PBX in the Cloud or utilising a Cloud based service offers many advantages:
- Reduced CAPEX
- Reduced in-house technical support requirement
- Access from anywhere
- Low cost & feature rich applications
However, there are risks too: we have to ensure that we don’t lose focus on delivering a technically effective solution and that we offer excellent voice quality at all times.
ISDN provides the benchmark for voice quality; indeed, it is still used for broadcasting. VoIP is too often known for poor voice quality, which might be acceptable for a conversation between friends on different continents, but is unacceptable for business.
Indicators of poor voice quality calls are:
- Broken audio
- Calls dropping
- ‘Underwater’ speech
- Calls failing to connect
- Echoes and crackles
- ‘Tinny’ sound
So what is the cause? And what can we do to provide ISDN-level quality when deploying a cloud-based VoIP solution?
Technical requirements for good quality VoIP traffic
There are four key technical requirements for good quality VoIP traffic, with the first often being the only one that is properly considered:
- Sufficient bandwidth & correct codec: VoIP generally uses SIP, with G711 being the industry standard codec. The advantage of this codec is that it doesn’t compress the call and provides ISDN call quality. G711 requires 88Kb per call on Ethernet and around 106Kb per call on Broadband, including overheads. Other codes such as G729 require less bandwidth but at the expense of call quality so are not recommended for business-grade voice quality.
- Low latency: Latency is the time it takes for a data packet to get from A to B. Latency higher than 150ms is audible and can cause cross-conversation.
- Low jitter: Jitter is the variation in latency, with voice requiring the packets to arrive in sequence and at regular intervals. A high jitter rate above 45ms will result in corrupt audio that is extremely noticeable and off-putting to the user.
- Low Packet loss: SIP traffic uses UDP rather than TCP. UDP is faster and better suited to voice, but lost packets are not resent. In order for a good quality call to happen, packet loss during the call needs to be as low as possible. High packet loss of more than 1% will be noticeable with words being cut and even missed entirely.
What is the cost of poor voice quality?
Poor call quality is not just frustrating to all concerned, it can be extremely costly through:
- Time wasted. Each poor quality call might waste 5-10 minutes in having to repeat the call or perhaps write an email instead. How do your customer’s employees feel about poor voice quality? We have had customers who have told us that they have lost key clients as a result of previous poor SIP implementations leading to call quality problems day after day.
- Lost business. If the poor quality call was a sales call, is there a chance that your customer is losing business as a result? What is the monetary value of each sale? £10, £100, £1,000? Perhaps more.
- Reputation. What message does poor call quality send to existing customers? However good your product and customer service, poor call quality can create a negative image of your company’s standing.
- Downtime. Worse than poor call quality is downtime. What are the risks of downtime? What is the cost to your business if your phones are not working?
a. Would you lose sales?
b. Would customers go elsewhere?
c. Would you miss deadlines?
d. Would your productivity drop?
e. Would staff morale fall?
To avoid downtime your cloud PBX service should be resilient and backed up fully with SLAs. Secondly, you must ensure that the connectivity to your service is resilient – not just to the Internet, but the entire route to your cloud PBX. This area is generally neglected!
How to ensure voice quality
Sadly, there isn’t a magic ‘QoS’ switch that you can flick to ensure voice quality end to end. Instead, you will need to look carefully at each component of the solution to ensure that voice quality has been considered from the start:
- The LAN: voice traffic should be protected to ensure it isn’t compromised by normal data traffic on the LAN. This is the easy bit, and there are two main ways to ensure it:
a. Create a dedicated voice LAN. This can be as simple as a dedicated voice switch into which the handsets plug via their own CAT5e/6 structured cabling.
b. Apply Quality of Service (QoS) on a converged LAN. Be careful to fully understand how this will work in both directions and ensure that other sensitive traffic is not compromised. If utilising softphone apps on mobile phones over a wifi network, remember to apply QoS here as well. However, this can have a tendency to be tricky and expensive.
- The access circuit: First, the circuit must support the technical demands of voice:
a. Bandwidth: remember, this is approximately 88Kb for Ethernet or 106Kb for Broadband per G711 call.
b. Latency: <150ms
c. Jitter: <45ms
d. Packet loss: <1ms
Secondly, as with the LAN the circuit should either be for voice only or, if converged, it must have a sufficient QoS policy enabled both downstream and upstream.
Sadly, much of the UK market seems to miss this point with the outcome being variable, and often poor, voice quality. This is primarily because wholesale broadband ADSL & VDSL circuits do not as standard offer any SLAs on bandwidth, latency, packet loss or jitter and do not offer bi-directional QoS to protect voice. Adding an upstream QoS policy on the customer’s router does little to combat this!
What you need
Our recommendations are as follows:
1 Fibre Ethernet, FTTC Ethernet or EFM Ethernet with appropriate QoS policy applied upstream and downstream.
2 SDSL M Broadband circuit solely used for voice. Our SDSL M circuits have sufficient performance SLAs for carrying up to 15 G711 voice calls.
3 This article is focused on voice quality, but don’t neglect security. Protect circuits or VLANs from potential hackers and expensive voice fraud by applying appropriate security measures. Spitfire runs a half-day course for our Partners on ‘Avoiding voice fraud’. It is not uncommon for delegates to take an unscheduled break in this course to make some quick changes to their PBX implementations!
4 Connecting to your cloud PBX provider. Cloud PBX providers will generally allow you to connect to them via the internet. Whilst this is flexible, it carries a major risk, as the internet is a contended public network that does not offer any consistency or guaranteed performance. A cloud provider might be top-notch for guaranteed performance, and so might your Ethernet circuit, but neither will take responsibility for issues on the public internet!
One solution is to install a dedicated access circuit from your premises to the cloud PBX provider. However, this is expensive and inflexible. Our recommendation is to use a single organisation for your ISP and Cloud PBX and check that they can offer a private route between your circuit and the PBX, thus avoiding the public internet.
5 Connecting to the PSTN network: SIP trunks are used to connect your cloud PBX service to the PSTN network so that external calls can be made and received. Most SIP trunks are delivered via an Internet connection so your calls will traverse the public internet for a second time – and we know the risks of that! To avoid this, ensure that the Cloud PBX provider offers its own SIP Trunking service so that calls do not route via the internet.
Voice quality can be assured by your Cloud PBX provider from the handset to the PSTN network by:
a. Configuring a Voice quality LAN
b. Utilising a Voice approved circuit
c. Implementing a Cloud PBX service that is directly connected to your ISP network
d. Utilising a Cloud PBX that uses directly connected, not internet SIP trunks
e. Ensure the G711 CODEC is used for high quality audio
f. Ask your providers if they can assure your voice quality end to end.
Want to know more?
If you would like to know more about how to ensure voice quality in the cloud, please come to Spitfire’s Seminar talk at UC Expo on 16th May at 12.20pm or visit Stand F105. Alternatively, please contact us on 020-7501 3150.
Meet us at:
– UC EXPO: May 16-17
– Margin in Voice & Data: June 14
– Channel Live: September 11-12
– IP Expo: October 3-4
– Offices in London & the Midlands
– Search ‘Spitfire Network Services’ on YouTube for videos on Spitfire SIP Trunks, Converged Ethernet, Spitfire Partner Service & much more.
for the following:
– The truth about VDSL and VoIP: why VDSL is not suitable for voice
– PBX Security in the VoIP Environment: how to avoid voice fraud.
Spitfire specialises in providing high quality Internet and Telecomms solutions engineered for, and tailored to, the specific requirements of our business customers. Our account managers are highly trained to ensure they gain a detailed understanding of data and applications through a training programme covering everything from coding to industry recognised networking qualifications such as Cisco CCENT and CCNA. This thorough training programme enables the account manager to accurately identify the network needs of our customers so that we can recommend the most suitable solution.