The Support of VoIP and Video Conferencing by IP Carrier Infrastructure

Executive Summary

Premiere Elect has main offices in Washington, London, Brussels, Moscow and Beijing; with associate offices in key cities around the world. Main offices tend to have six staff. Associates are independent organisations brought in as required from preferred contacts. Voice and video communications are needed between main offices, main offices and associates, and main offices and clients. Premiere Elect PR has engaged an IT consultant to address its networking needs. The consultant has realised that there are several VoIP technologies available that would meet the company’s needs, but needs your help to identify the most appropriate.

Within this written report I will be comparing and detailing two voice and video based communication applications, Cisco Jabber and Microsoft Skype in order to establish suitability between main offices and associates, and main offices and clients. Premiere Elect PR has engaged an IT consultant (myself) to address its networking needs. I will be discussing positives and negatives of VoIP Technologies whilst discussing other issues such as; what support VOIP and video conferences get by IP carrier infrastructure, the effects of jitter and delay will have on everyday situations, what Codecs are used for both applications and what protocols or standards each service follows.

Introduction

VoIP (voice over IP) is the transmission of voice and multimedia content over Internet Protocol (IP) networks. VoIP uses codecs in order to encapsulate audio into data packets, transmitting the encapsulated packets across an IP network and the UN-encapsulate packets back into audio form at the other end of the connection. VoIP can reduce network infrastructure costs, subsequently it enables providers to deliver voice services over their broadband and private networks, and allows enterprises to operate a single voice and data network. (TechTarget, 2019)

(Figure 1A: Nextiva, 2016 VoIP Advantages & Disadvantages)

Cisco Jabber and Skype are two of the most widely used conferencing applications in the world. Both applications are developed by highly reputable companies with extensive background knowledge on business applications and hardware. Cisco (Jabber) is a popular networking, video conferencing and office equipment manufacturer while Microsoft (Skype) is the pioneer behind Office applications.

Cisco Jabber offers excellent features allowing its users to IM and presence without any hassle and shows the availability of all your contacts at a glance, Jabber allows for instant messaging chats to be initiated with an individual or group without any delay. Cisco is also compatible with both internal and external communication.

Cisco Jabber offers optimal communication across numerous platforms and devices much like Skype for Business. It lets you access presence, instant messaging (IM), voice, video, voice messaging, desktop sharing, and conferencing. Some of the features offered by Cisco Jabber:

Some of the features offered by Cisco Jabber:

• High-quality video and audio calls.
• Presence and Contact Information (Reduce Communication Delays)
• Cisco WebEx compatibility when in a meeting or sharing an application.
• Contact Directory
• Point-to-point chat with co-workers inside your network, or supported federated business and personal contacts.
• Group Chat
• Multiple chat rooms

(Ctelecoms.com, 2017)

Skype is an instant messenger service application that offers a VoIP (‘Voice over Internet Protocol’) service. This lets you make calls to people all over the world via your computers or telephone and have text chats. Calls between computers are usually free, and other Skype calls are generally inexpensive. Skype is primarily free to use and is available across multiple platforms, allowing allows users to make Skype calls using a desk phone, mobile device, smart tv or computer. Although this feature is only available to organizations with use of a Skype Server.

Skype for Business allows its users to make Skype calls through a desk phone. Although this feature requires a Private Branch Exchange (PBX) desk phone which is configured to work with Skype for Business. This allows for users to search within the skype interface before selecting and initiating the call on their desk IP phone.

Skype for Business is owned and designed by Microsoft who subsequently own Office related applications, this allows for easy accessibility when perhaps initiating an instant message or voice call with a contact seen within an email. Office also uses skype’s user status within its applications, showing when a user is online this can often aid in email response times.

Some of the features offered by Microsoft Skype:

• Cost Saving (Free to Download)
• Cross-Platform
• Presence and Contact Information (Reduce Communication Delays)
• Rate My Call (Troubleshooting & Feedback to Developers)

(DigitalUnite.com, Unknown)

The support of VoIP and video conferencing by IP carrier infrastructure –20%

The quality of the VoIP Infrastructure can be influenced by a number of variables for example its financial impact, Scalability, Accessibility and Security. A carrier network is the proprietary network infrastructure belonging to a telecommunications service provider such as BT, Sky, Virgin and TalkTalk. Carrier networks are made up of large, complex configurations of hardware, interconnected to provide communications services over large geographical areas. (TechTarget, 2019)

Skype is a multipurpose client that provides voice communication features as well as standard instant messaging features, such as text messages and file transfers. A prime example of its infrastructure is the three main types of computers used within the Skype service: a standard node, a super node, and the Skype server. The main feature of Skype is Voice over Internet Protocol (VoIP), where users can communicate via voice. (ScienceDirect.com, 2005)

According to (TechTarget.com, 2016) ‘In a communications network, a network node is a connection point that can receive, create, store or send data along distributed network routes. Each network node — whether it’s an endpoint for data transmissions or a redistribution point — has either a programmed or engineered capability to recognize, process and forward transmissions to other network nodes.’

Client server and Peer to Peer (P2P) model are two of the most common methods used. Skype depends on the usage of super nodes in its network which form a peer-to-peer (P2P) network. Skype doesn’t rely on the usage of a single computer and therefore will usually hold two different groups, standard nodes and super nodes, super nodes are grouped based on the level of spare bandwidth and public locality. Skype is moving away from P2P primarily due to its growth, making it easier to manage the network. Battery efficiency on mobile devices is another reason for the migration, mobile users have been exposed to significant power usage when browsing the application, as well as the requirement to use more of their mobile data. All of which were subsequently incurring costs. (Lifewire, Unknown) Figure 1.B Details the topology view of the skype network infrastructure

The Effects of Jitter, Delay and Packet Loss –20%

Latency is defined a measure of delay, measuring the time it takes for data to get from source to its destination across the network. Latency is usually measured as a round trip delay – the time taken for information to get to its destination and back again.  A round trip delay is an important measure of data as a computer that uses TCP/IP, sends an initial amount of data to its destination and before awaiting an acknowledgement prior to sending any more.  Typical, approximate, values for latency are as follows:

• 800ms – Satellite
• 120ms – 3G cellular data
• 60ms – 4G cellular data which is often used for 4G WAN and internet connections
• 20ms – MPLS network such as BT IP Connect, when using Class of Service to prioritise traffic
• 10ms – Ethernet Network such as BT Ethernet Connect or BT Wholesale Ethernet in the UK

(sas.co.uk, 2019)

According to Cisco’s website Jitter is defined as a variation in the delay of received packets (Cisco, 2006). High jitter results in choppy voice or temporary glitches. VoIP devices implement jitter buffering algorithms to compensate packets that arrive at high timing variations, and packets can even get dropped when they arrive with excessive delay, resulting in no transmission on the target end. (NetBeez, 2016)

Testing the quality of VoIP has become easier and services have greatly improved over the last few years due to both the providers becoming more reliable and the ISPs offering better connection.  This advancement in the quality of services has helped increase the number of VoIP subscribers, but occasionally issues affecting voice quality do arise. VoIP calls often are in the 3.5 to 4.2 MOS range.  Figure 1 can be used as a guide for VoIP MOS testing and a good comparison for voice quality. (VoIPMechanic.com, 2015)

Any form of packet loss can prove to have a significant impact on VoIP applications, VOIP is not tolerant of packet loss. Even 1% packet loss can “significantly degrade” a VOIP call or connection between peers. “The person on the other end sounds Robotic” (community.spiceworks.com, 2013) is a common result of packet loss and can result in partial or all audio transmission being distorted.

Packet Size is an important factor when it comes to the transmission of VoIP data, High packet size requires a lower bandwidth as fewer packets will be required during the transmission, however if the packet size is excessive the overall encoding and de-coding duration could create un-needed latency. On the other end, smaller packet size transmission would require the utilisation of more packets, subsequently a faster internet connection with greater bandwidth would be required to prevent any delay.

The following techniques can be used to reduce the effects of latency, jitter and packet loss:

QoS (Quality of Service) is a configured function which determines the level of traffic to each device, service and IP. By using QoS within a VoIP environment, this guarantees the majority of all packets will be subject to use by the VoIP service only. Secondly the Implementation of a Jitter buffer will store the data packets before intentionally delaying the arrival of each packet(s) so that the end user experiences a clear connection with very little sound distortion. There are two kinds of jitter buffers, static and dynamic. A static buffer is hardware-based and manufacturer issue. A dynamic jitter buffer is software-based and configured by the network administrator.

(TechTarget.com, 2008)

Speech Encoding –10%

A Codec is a term for the following variations, compression and decompression. A Codec is embedded within a device such as a mobile phone or VoIP phone. In the telephony environment, a codec is sometimes used as a mechanism to maximise bandwidth utilisation by compressing the data of the call and then decompressing it when the call is delivered.

(Figure 2: SciAlert.net, 2003 Speech Encoding System)

An example of a popular codec is G.711, There are two varieties of this Codec, namely U-law and A-Law. U-law is typically utilised within North America and Japan, whilst A-law is typically used by the rest of the world. G.711 is the desired codec when wanting the best call quality from a VoIP System. G.711 does not use any form of compression, and as a result, the call quality sounds like using a regular ISDN phone. This Codec is supported by most VoIP providers. (Nexbridge.co.uk, 2016)

G.722 is another characteristic of the wideband audio coding system and is mainly used in VoIP such as on local area networks, where network bandwidth is easily available and offers improvement in speech quality over narrow-band codec such as G.711, without an increase in implementation complexity. G.722 is also used by broadcasters for sending commentary-grade audio over single 64 Kbps integrated services digital network B channels. G.722 VoIP is carried in Real-Time transport protocol payload types and uses sub-band adaptive differential pulse code modulation with bit rates of 64 Kbps and is referred to as 64 Kbps audio coding. (Techopedia.com, N.D)

G.729 is an alternative codec which is known to offer a good level of call quality at a low bit rate of 8Kbps (kilobits per second), which would mean that you would be able to get more calls through your bandwidth that if you were to use the G.711 Codec. The G.729 code utilises a small amount of bandwidth when transmitting data the call data, however requires higher CPU utilisation, subsequently VoIP devices may only able to handle one call at a time. (Nexbridge.co.uk, 2016)

Opus is another popular codec used within a wide range of audio applications, Opus is a totally open, royalty-free, highly versatile audio codec. Opus is unmatched for interactive speech and music transmission over the Internet, but is also intended for storage and streaming applications. It is standardized by the Internet Engineering Task Force (IETF) as RFC 6716 which incorporated technology from Skype’s SILK codec and Xiph.Org’s CELT codec.

Opus can handle a wide range of audio applications, including VoIP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bitrate narrowband speech to very high quality stereo music. (opus.codec.org, N.D)

SILK is an audio compression format and audio codec developed by Skype Limited. It was developed for use in Skype, as a replacement for the SVOPC codec. Since licensing out, it has also been used by others. It has been extended to the Internet standard Opus codec.

SILK codec is offered as a real time implementation that can be configured to support multichannel wideband audio VOIP applications. SILK operates at four different sampling rates: 8 kHz narrowband, 12 kHz medium band, 16 kHz wideband, and 24 kHz super wideband. These sampling rates allow for the capture of higher frequencies, which provide fuller sound, while also allowing interoperability with the Public Switched Telephone Network (PSTN)

SILK voice codec has three complexity settings: low, medium, and high. The complexity settings are used for pitch estimation and noise shaping and the number of stages used in the quantization of the noise shape parameters, the Long Term Prediction (LTP) coefficients, and the Normalized Line Spectral Frequencies (NLSF). SILK subsequently takes advantage of today’s powerful processors while not abandoning older processors.

(Vocal technologies, 2017)

Standards and Protocols Explained–25%

Even though VoIP is completely different than traditional telephony, phones still have to ring, numbers must be communicated, and routes have to be set up, these functions are handled by the signalling protocol. The three most known protocols are H.323, Skinny, and the Session Initiation Protocol (SIP).

Session Initiation Protocol (SIP) is a signalling protocol used when initiating, maintaining, modifying and terminating real-time sessions involving video, voice and other multimedia communication methods between two or more endpoints on an IP network. (TechTarget.com, 2008)

SIP was developed by the Internet Engineering Task Force (IETF) drive the evolutionary needs of IP-based communications. Native support for mobility, interoperability and multimedia were among the drivers behind SIP’s development. SIP complements other communications protocols, such as Real-Time Transport Protocol (RTP) and Real-Time Streaming Protocols (RTSP), used in IP-based sessions. (TechTarget.com, 2008) SEE FIGURE 2.

There have been previous standards and protocols like Speex, Skinny, and H.323 that have been VOIP protocols however SIP is the de facto standard. Legacy protocols are also used over IP networks such as SIGTRAN and ISUP but more frequently these are now merging across into the SIP domain.

Skinny Client Control Protocol (SCCP) is proprietary control and communication protocol taken over by Cisco Systems although developed by Selsius Systems. Skinny is a lightweight IP-Based protocol used for communication between Cisco Call Manager and Cisco VOIP phones.

H.323. is a legacy known protocol within VoIP communication, Both SIP and H.323 aim to achieve the same end result. H.323 not need to be present on a network, which is perfect for enclosed private networks. This would be a bad fit for a global business. H.323 is recommended by the ITU Telecommunication Standardization Sector (ITU-T), which defines a set of standards for the transmission of packet multimedia data over networks.

(TrueConf.com, 2018)

Conclusion

Skype for Business and Cisco Jabber are very similar and both offer a high standard when providing Instant Messaging, Voice & Video Calls, Voice-Mails and application integration(s). The best application is down to the individual business requirement(s). Microsoft and Cisco are both reputable companies who are well established within the networking and communications sector.

Cisco, holds decades of experience within the hardware based video conferencing and the networking infrastructure where as, Microsoft is well known for its software based video conferencing and business applications

Based on Premiere Elect’s requirements I have chosen Microsoft’s Skype, based on the user accessibility and overall application functionality, Skype for Business is not only providing a cheaper alternative to PSTN but it’s providing a revolutionary service.

Figure 1: Mean Opinion Score for VoIP Testing Table

Figure 2: SIP Diagram – Peer to Peer
https://www.voipmechanic.com/sip-basics.htm

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