VoIP Now

Digging Into The Skype Protocol

Nuno at 21Talks reveals a bit about the inner workings of Skype. I've speculated before, but he confirms that Skype uses UDP (User Datagram Protocol), a connectionless protocol, instead of TCP (Transmission Control Protocol). This backs up my previous speculation as to why Skype sometimes has crappy voice quality: data packets can arrive in which order they want to. TCP, on the other hand, orders packets going in and coming out. So if there are network problems, Skype will exhibit the "warbled" voice phenomenon. This could also happen if free RAM and/or processing cycles on your computer are low, resulting low quality VoIP calls.

It might be due to the holiday season, but my cable connection has been especially horrible for over two weeks now. While I've only used Skype for VoIP during this period, I'm hazarding a guess that any VoIP service is suffering some sort of problems.

VoIP Roundup - Wed Nov 15/06

Skype Enterprise Features Coming?
Skype execs have hinted at upcoming enterprise and call center features. So maybe this will be how eBay finally monetizes Skype?

Speed Demons
The 100-Gigabit Ethernet (100-GbE) technology is here, being demonstrated by a number of companies and the University of California Santa Cruz. A test run sent a signal from Tampa, Florida to Houston, Texas, and back - a first for a live production network. If I understand this correctly, IP backbones will get this technology fairly soon. And as 100-GbE becomes commonplace, likely in several years time, it should mean some incredible real-time video conferencing ability, superfast downloads of movies, and live video broadcasts, to name just a few benefits.

Legal Issues Surrounding VoIP Enterprise Implementations
TechRepublic details legal issues to be aware of when planning a VoIP implementation. They have real alphabet soup of issues, some of which I've only peripherally aware of: SOX/ Sarbox (Sarbanes-Oxley Act), GLBA, HIPAA, E911.

Wibree: New Nokia Wireless Connectivity Protocol

Nokia has introduced Wibree, a short-range wireless connectivity protocol, which is complementary to Bluetooth and UWB. It's the culmination of five years of development and is ten times more energy efficient than Bluetooth. Wibree operates in the 2.4Ghz band, has a range of 10 meters, and a maximum speed of 1Mbps (megabit per second). (Bluetooth operates in the UHF above 6 Ghz.)

Instead of being used in cell phones, it would be in watches and possibly enterprise devices such as keyboards, mice and digital pens. Other possibilities are "wearables" such as intelligent jewelry. Nokia already has dula-mode cellular VoIP phones. I'm wondering if Wibree could be used for VoIP "communicator" badges that double as lapel pins or brooches - similar to Vocera's Call Badge.

[via CBR Online, CRN]

VoIP Roundup - Mon Sep 25/06

Universities Banning Skype
A number of universities have decided to ban Skype, stating that it consumes bandwidth and supposedly is an "illegal" waste of resources. (Illegal? Seriously?) Grid computing apps are also included in the ban at several California universities including University of California Santa Barbara, San Jose State University, and California State University Dominguez Hills. They are not banning Gizmo Project or Wengo. [via Ars Technica] Several countries also ban/ block Skype, including Korea and the UAE (United Arab Emirates).

Telrex CallRex VoIP Call Recording For Cisco
Telrex claims that their CallRex version 3.1 is the  first VoIP call-recording solution to be certified for encrypting Cisco Unified CallManager 5.0 calls. [via Business Wire]

SIP Trunking Makes VoIP Telephony More Flexible
Rich Tehrani reflects on how SIP trunking has made IP telephony more flexible by reducing the amount of proprietary hardware. He points out that not all IP PBXes are connected to SIP trunks; that over half of them (some used by IP-based call centers) are still using PSTN trunk lines.

Is SIP The Building Block For IP Telephony Features?

Many people think that the SIP-based technology will be the core building block for future enterprise IP telephony networks, especially for call control protocols and offering features for VoIP that are already present in existing telephony systems.

SIP, or Session Initiation Protocol, is an IETF proposed standard to manage online multimedia sessions that include video, voice, IM, and more. As such, there is expected to be strong interest in the SIP market in the near future. [via CNS Magazine] Light Reading has published a new report studying the SIP market  (US$900).

VoIP Roundup #3

Michael Kanellos sums up the value of using RebTel's VoIP service very nicely: instead of losing your unused monthly cell phone minutes, use them in international or other long-distance calls using RebTel. They create a bridge service by creating a dedicated local number that saves the caller long distance charges. For $1/week, that's not bad if you tend to put off long-distance calls because of cost.

BT (British Telecom) has been laying out plans to get into the internet telephony business (i.e., VoIP). They're looking to have one million internet phone customers in the next year. The company was formerly a monopoly and originally concerned about VoIP, but has changed its stance.

A Taiwanese government web page says that Taiwan is preparing for a VoIP explosion. The output value of VoIP-related products rose to US$460+ million in 2005, up from about US$370 mln and US$262 mln in the previous two years. A government-funded research institute has set up the ViTA (VoIP Taiwan) Forum to set up VoIP product certification standards in that country.

Arcosoft is offering VONaLink SoloRecord recording software for SIP-based VoIP phone systems, including Vonage. Either soft or hard VoIP services are supported. Calls are recorded in .WAV or .MP3 format, and an audio watermark can be audio to prove non-tampering.

Robert Poe of VoIP News shows why IP PBXs are going to replace their TDM counterparts in the enterprise.

Network Inter-compatibility - A VoIP Holy Grail?

According to comScore [via TechCrunch and others], it appears that the Google  Talk IM does not have all that many users. Google Talk is a distant fourth in the list of IM clients. The top three spots are taken by MSN Messenger, Yahoo Messenger, and AIM. We're not talking total number of downloads, just number of users.

These numbers indicate that, as of May 2006, there are only just over 339 million IM users of all flavours, out of over 900 million Internet users worldwide. MSN is at the top with about 60% of the market and Google Talk has only 1%.

The numbers appear to be for text IMs only. Consider that for Skype, I've been numbers like 250 million (PDF, 1 page) and 280 million downloads, and 100 million registered users (which was passed in early May 2006). Google is nowhere close to neither Skype nor the top text IM clients. Note: the TechCrunch article points out that the comScore numbers do not include the embedded version of GTalk within GMail. Some GMail users apparently prefer to use the native IM.

Some of the above IMs (in fact, possibly all - I don't know much about AIM) have direct VoIP (PC-to-PC) capability. Windows Live Messenger and the latest Yahoo! Messenger now have VoIP ability as well.

Regardless, the numbers suggest that Skype has to be able to keep up, especially with the announced IM alliance between Microsoft and Yahoo! When their respective IM's become compatible, together they'll have over 83% of the IM market (using current numbers).

With that kind of market share, don't have to swallow their pride and use the open source Jabber/XMPP, but it sure would be nice. That's what Google, Trillian, and several other lesser-known IMs use. Can you imagine an Internet where you can choose your fave IM/ VoIP soft client and connect to anyone? Skype, MSN, Yahoo, AIM, Google Talk.... Nice.

As for Michael Arrington's suggestion (at TechCrunch) that Google come up with a strategy to grab more market share, I recommend they buy Vozin Communications. Their Talqer soft client turns Google Talk into a true VoIP client.

From my point of view, once all the technical kinks are worked out of VoIP services, including e911 and improved call quality, every person on Earth who uses a phone will be using a straight or hybrid VoIP phone or soft client. If they were to all become intercompatible, then these usage numbers might be perceived as meaningless - especially by the marketing engines of the larger companies. This would be a reason for them not to work towards compatibility. Too bad, because we could use more IM/VoIP network intercompatibility.

Aside: If you do a Google Search for "how many Skype users worldwide", you can see how fast Skype downloads increased in 2005. Or you could read Jean Mercier's post at SkypeJournal (about downloads in the US and Canada), which suggests that the number of downloads needs to be filtered for previous users who were upgrading one or more times since they first downloaded Skype.

Bell Canada Personal Communication Manager introduced

Bell Canada, Nortel and New Heights Software have introduced the bell Canada Personal Communication Manager system (PCM) which is a PC based soft client combo of voice, multimedia communications and messaging which has been integrated into a single interface that can be used in office or while working remotely. 

The PCM offered by Bell Canada enables laptop users to undock their machine and carry with them the same telephone and messaging service and profile which can be used at their corporate desktop. It also enables users to initiate calls and multiparty conference sessions can be conducted with just a point and click. 

Via [VoIP for SMB]

T-Mobile Bans VoIP from Its 3G Network

T-Mobile has banned VoIP from its super 3G data network. The reason sighted by the company is poor quality. Earlier it was said that the move was a commercial decision but in a statement issued on Wednesday, the company stated that VoIP technology was not of high level quality so that it could provide a good customer experience on the T-mobile network.

The company also stated that there might be a change in the situation in the future but as of now in the interests of its customers it was necessary top restrict the use of VoIP technology.

Via ispreview

US VOIP suppliers face obscure export regulations

Did you know that the section 740 of the export control regulations contains a definition that restricts companies from selling products in other countries that can support "concurrent encrypted data tunnels or channels exceeding 250" connections at one time?

That's peanuts compared to the the capabilities of modern networking products that routinely handle thousands of simultaneous "channels" or users.  The obscure piece of legislation can hinder say, Cisco's export of advanced networking products.

However, companies can easily sidestep the legislation and easily develop the products overseas

Via Networking Pipeline

New Law grants VoIP Providers Access to E911 Systems

The US House of Representatives Subcommittee on Telecommunications recently passed a bill which gives all VoIP service providers access to critical Enhanced 911 (E911) infrastructure. The bill grants access to selective routers, databases, numbering resources and other essential elements for the provision of E911 for nomadic VoIP services.

Access to E911 is a major worry for VoIP providers like Vonage, all this while. The passing of the bill ultimately benefits the consumer. Moreover, all voice service providers can also access the infrastructure they need to offer enhanced 911 to their customers.

Via Yahoo Finance

ShoreTel scores perfect ten in a study conducted by Nemertes Research

According to an independent study conducted by Nemertes Research, ShoreTel Inc has been given full marks for its value, troubleshooting, easy installation and overall performance. This is for the third year in a row that IT professionals have ranked ShoreTel ahead of companies like Cisco, Avaya and Nortel. This year it beat those companies by a bigger margin as compared to last year.

According to Robin Gareiss, Nemertes’ executive vice president and senior founding partner:

It’s rare to have a single vendor dominate ratings in any technology category for three years in a row.

via  [TMCnet]

Speech Engine receives Best of Show award at Internet telephony Conference and Expo East 2006

LumenVox’s Speech Engine has been awarded the Best of Show award at TMC’s Internet Telephony Conference and Expo East 2006. Speech Engine offers developers with a flexible API that performs recognition on audio data from any audio source. It includes server side grammars and MRCP and Linux support. The innovative Speech Engine is an indication of why so many enterprise buyers, resellers, developers and service providers flock to Internet Telephony Conference and Expo.

"LumenVox and their innovative Speech Engine are a standout indication of why so many enterprise buyers, developers, resellers and service providers flock to Internet Telephony(R) Conference & Expo," said TMC President and Conference Chairman, Rich Tehrani.

via  [VoIPMagazine]

Teles files lawsuit against Nokia for patent infringement

Teles AG has filed a lawsuit against Nokia for alleged patent infringement. The company claims that Nokia’s 6136 handset and its GSM fallback function infringes Tele’s European and German patents. The case has been filed with the Mannheim regional court. Nokia has touted the new device at the 3GSM world Congress in Barcelona.  The phone makes use of unlicensed Mobile Access technology which makes use of GSM/GPRS voice and data services over a broadband connection. The company plans to make the 6136 available by the second quarter of 2006.

via  [TMCnet]

Ericsson and VimpelCom become the first to demonstrate 3GPP standard based compressed speech and signaling over IP

Ericsson and VimpelCom have become the first company to demonstrate 3GPP standard based compressed speech and signaling over IP in a commercial GSM network. In December 2005, first call was made between Yekaterinburg and Nizhniy Tagil in the new VimpelCom coverage area. It is a state of the art technology which would be launched across the whole region in February 2006. Ericsson’s VP (Market Unit Eastern Europe and Central Asia) Zoran Lukovic says that their company is proud to be selected by VimpelCom for installing softswitch solution and supporting its evolution to all IP.

via  [VoIPCentral]

Keynote Systems releases study on consumer VoIP quality

Keynote Systems has released a new study on consumer VoIP quality. The report states that although the reliability has improved but the audio quality is far behind the traditional phone service. The testing was done from mid November till the end December 2005. During this period eleven services were evaluated. Time Warner Digital Phone emerged as a winner in terms of both audio clarity and reliability while Verizon’s DSL emerged as the most reliable network. Three VoIP providers reached a toll quality audio which exceeded MOS score of four out of five. One of the major areas of concern is that there has been no considerable change in the amount of audio delay present in VoIP calls over the past six months.

via  [Von Magazine]

Extreme Networks emerges out winner in benchmark performance test conducted by Tolly Group

A recent benchmark performance test conducted by Tolly Group proved that Extreme Networks’ BlackDiamond and Summit switches performed better as compared to other switches when the network was required to protect prioritized voice traffic running on an Avaya IP Telephony application. The core/edge LAN infrastructure’s ability of Extreme Networks to deliver triple play voice, video and data applications in networks with varying levels of congestion was evaluated by the Tolly Group. It was found out during the test that Extreme Networks’ BlackDiamond and Summit switches delivered hundred percent of VoIP calls and toll quality voice when low priority background traffic load increased.

via [ prnewswire ]

GNU Telephony stack

The GNU Telephony stack is an alternative for those who do not wish to use proprietary VoIP stacks. Tycho Softworks is backing the GNU Telephony stack. GNU Bayonne, which is one of the major components of the stack, is maintained by the owner of Tycho Softworks. The Free Software Foundation sponsors GNU and the GNU Telephony stack consists of an elaborate list of applications sponsored by GNU. internetnews.com reports:

The Telephony stack includes the GNU Bayonne telecommunications application server, the Troll ip/pstn gateway packages, the GNU RTP (define) stack, the Open H.323 stack and the GNU Gatekeeper H.323 call server.

Read More: GNU Telephony Stack Opens Up VoIP

Selecting the right standard

In order to implement a VoIP network, it is important to select the appropriate standard. A number of standards have been developed by the different standards bodies. These standards cater to the needs of different telephony environments such as Enterprise IP telephony, carrier long distance, call center, residential class 5 switching, etc.

Businesses need to understand the peculiarities of each environment and select a standard that will minimize the need to redesign the network and prevent the applications from getting outdated quickly.

In the early days of VoIP, several standards were developed for the sole purpose of integrating VoIP with PSTN. Even today, major VoIP carriers connect VoIP calls using PSTN. This is because VoIP peering still has to grow to a scale, which will allow users to bypass PSTN. tmcnet.com reports:

Computer Telephony Integration may not appear to be a critical design choice, but in 1995, neither was email. No doubt, the gap between computer and telephone will close over the next few years.

Read More: Which VoIP Standard to Use?

Jingle by Jabber

The preliminary documentation of Jingle has been published by the Jabber Software Foundation (JSF). Jingle is a set of extensions to the XMPP developed by the IETF. XMPP can be used for P2P multimedia sessions such as VoIP. lightreading.com reports:

Jingle provides a powerful framework for peer-to-peer multimedia sessions," said Peter Saint-Andre, Executive Director of the Jabber Software Foundation and co-author of the Jingle specifications.

Read More: Jabber IM Adds VOIP

Training for VoIP

According to a survey by Foote Partners VoIP along with storage/storage-area networking and Gigabit Ethernet was among the three most paying network related skill for the period July – September. Training for VoIP is basically of two types. The first type is the vendor-specific training that may be obtained by enrolling for a certification course that is managed by one of the many vendors such as Cisco, Avaya, etc.

The second type of training is not vendor-centric and may not result in a certificate from a vendor but it gives the learner an all-round understanding of VoIP and issues related to its deployment. This type of VoIP training is provided by TRA, the Teracom Training Institute, and Global Knowledge.

CompTIA, which offers vendor-neutral training, is endeavoring to offer certification to the students. It is in the process of developing its Convergent Technologies Certification. Vendor-neutral training offers IT executives and management personnel, who are involved in executing VoIP initiatives, the opportunity to learn about VoIP architectures and deployment-related issues.

Five nines for IP telephony systems

With the advent of VoIP, the conventional approach to calculating availability which was developed during the days of PSTN, does not hold true. VoIP systems are more distributed and are capable of supporting parallel structures in a manner not possible with PSTN. They can also include greater redundancy.

This can lead to high availability, which has to be arrived at after considering all the elements of the distributed architecture, such as the service components, software, hardware, etc. An availability of 99.999% or five nines implies a downtime of less than one hour every decade. spanlink.com reports:

VoIP-based IP Communications solutions are different. Intelligence is distributed to devices throughout the enterprise network: routers, firewalls, media gateways, LAN switches, IP phones, the IP PBX, and so on. Each device is itself a complex hardware/software system.

Read More: Taking VoIP Beyond Five-Nines

ICE receives support

Interactive Connectivity Establishment (ICE), which is a standard developed by IETF, has received support from Cisco Systems and Microsoft. ICE is used as a NAT traversal solution. The technology is considered to be more robust and scalable as compared to tunneling that is done using HTTP and Port 80.

Wi-FI Alliance to certify phones

The Wi-Fi Alliance will be certifying features meant to increase the battery life of LAN phones. The Wi-Fi alliance also offers certification to interoperable Wi-Fi products. It will now offer a label referred to as the Wireless Multimedia (WMM) Power Save. The label will serve as an identification of products that have succeeded in reducing the power consumption for multimedia applications over LAN. pcworld.idg.com.au reports:

The size and weight requirements of Wi-Fi phones as well as dual-mode cellular and WLAN handsets, plus the need to carry them around all day, make power consumption even more critical.

Read More: Wi-Fi group to certify battery-saving tools

MEGACO/H.28

The MEGACO/H.28 standard was developed by the IETF and the ITU-T. It is used for facilitating communication between the media gateway and the media gateway controller. It has a distributed gateway architecture and it assumes that the intelligence for processing calls is present in the media gateway controller and that access to the media streams is managed by the media gateway. As against MGCP in which the commands are applicable to the connections, in MEGACO the commands are applicable to the Terminations that are related to a Context.

The Termination, which can also be multimedia, sources and/or sinks either single or multiple media streams. The Add, Subtract, and Modify commands can be used to alter the Contexts. A Connection is required when two or more than two Terminations are placed in a common Context. The MEGACO commands include Add, Modify, Subtract, Move, AuditValue, AuditCapabilities, Notify, and ServiceChange. Unlike MGCP that is defined specifically for UDP/IP transport MEGACO supports UDP/IP, TCP/IP, or ATM.

IAX

Analog telephones are easy to deploy and use as they already have fixed standards and an infrastructure in place. This is not the case with IP telephony. VoIP protocols are not very easy to configure and this hinders the smooth deployment of VoIP phones.

IAX is a protocol created with the objective of reducing bandwidth consumption for signaling and facilitating transparent NAT. IAX employs UDP instead of RTP over Port 4569 for the transmission and receipt of signals and media. IAX enables easy traversal of firewalls with reduced overhead. It uses binary-only data instead of parsing text commands.

IAX devices are apt to recognize a dead line quickly because the protocol responses do not have to negotiate a foreign IP address. They are returned directly to their point-of-origin. A Layer 2 data link layer is used for signaling. The header size of the audio packets is not more than 4 bytes, this helps to conserve bandwidth. IAX trunking combines the data from several channels into one packet and reduces the number of headers as well as the number of packets. This attribute is useful with wireless networks.

An analog terminal adapter is used for implementing the IP stack, IAX, stack, TDM interface, echo cancellation, and caller ID generation. The IAX ATA device can convert an analog phone into a VoIP phone. The acceptance of IAX is expected to increase with a documented standard. The protocol is also going to include encryption and intercom functionality.

FACT-SIP

Motorola plans to integrate its FACT-SIP software package with MicroTCA and AdvancedTCA hardware in order to develop VoIP Open Application-Enabling Platform families. The ComStruct packet voice resource hardware by Motorola uses the FACT-SIP, which sends SIP commands across an IP socket. This facilitates the creation of VoIP-enabled applications like VoIP access gateways.

Motorola is focusing on integrating packet voice resource boards with SIP software to develop voice-enabled SIP applications that does not require low-level code. FACT-SIP consists of SIP protocol software and a management interface that enables the reconfiguration of a packet voice resource board by using a web browser. FACT-SIP is expected to be available by the first quarter of 2006.

MGCP

The physical and logical sides of a softswitch need to be interconnected in order to establish communication between the media gateway and the media gateway controller. The media gateway protocol sends commands to the media gateway under the master/slave arrangement by using two protocols. One protocol is the MGCP, which was developed by the IETF and documented in RFC 2705; it is documented in its updated version in RFC 3435.

Gateways that can employ this protocol include trunking gateways present between telephone and VoIP networks; voice over ATM gateways; residential gateways that provide a RJ11 interface to a VoIP network; and other such gateways. The information between the media gateway controller and the media gateway can be categorized as either events or signals. The two categories are supported by endpoints such as telephone or video systems. MGCP commands are used for used for the communication between the call agents and the gateways. The commands include EPCF, RQNT, NTFY, CRCX, MDCX, DLCX, AUEP, AUCX, and RSIP.

Voice over Wi-Fi with SIP

SIP facilitates the deployment of Wi-Fi by companies as it provides more alternatives in terms of hardware and other technologies that mobile users can avail.

With Wi-Fi deployments, companies do not have the options that are available with the wired VoIP implementations where client devices provided by the vendor enable interoperability. Even though SIP enables companies to compare products from a number of vendors, it is possible that these phones may not have certain features provided by proprietary phone solutions. However, dial tone, caller ID, redial, transfer, etc are core features that are present in SIP-enabled phones.

SIP can be used for more than just voice services because it is a signaling protocol used for call setup and teardown and is not involved in the actual delivery of the content, which can occur in a peer-to-peer fashion. Push-to-talk and MMS are just two of the many applications that can make use of SIP. The technology that enables the establishment of user presence by using SIP is explained in RFC 3586. SIP allows mobile users to register with the SIP server and obtain connectivity regardless of the network. Thus, SIP can facilitate roaming between cellular and Wi-Fi networks.

MGC

Telephone calls made using either PSTN or a VoIP network need to have end-to-end reliability and the establishment and severing of end-to-end connections should be possible.

A fair amount of network intelligence resides in the Media Gateway Controller (MGC) as it enables the above mentioned functions. The MGC provides call routing function and control of the connection and the network resources. The execution of these functions involves activities such as managing the origination and termination of signaling messages between end user stations, external networks, etc. The MGC also maintains call state information for calls on the Media Gateway. It also functions as a channel for negotiating media parameters such as audio/video codecs that may affect bandwidth consumption.

Ports on the Media Gateway and the network bandwidth availability are managed by the MGC. MGC manages access capabilities and permissions for endpoints. According to the IPCC architecture, there are two additional functions meant to be performed by the MGC. These include the Call Agent Function and the Interworking Function. The call control and call state maintenance operations are provided by the Call Agent Function. This function uses the SIP or H.323 protocol among other protocols. The Interworking Function is used to provide H.323/SIP or IP/ATM network connections.

IPCC

The International Softswitch Consortium was founded in 1998 to further the deployment of next generation switching and VoIP technologies. The consortium is now known as the International Packet Communications Consortium (IPCC) and covers video over IP and services accessed through wireless and wireline networks. The IPCC works to facilitate VoIP and softswitch deployment by utilizing the services of service providers, system integrators, etc who define reference architectures and management systems.

One of their key contributions of IPCC is the Reference Architecture, available here. There are four functional planes referred to as Transport, Call Control and Signaling, Service and Application, and Management. These define the functions of a VoIP network.

The transport plane provides functions such as call setup and call signaling. The technology used for ferrying the media may vary. Packets are transported across the VoIP network using IP Transport Domain that holds devices like routers and switches. Signaling gateways, media gateways, and interworking gateways make up the Interworking Domain, which interacts with networks outside the VoIP network. Access or residential gateways connect non-IP terminals such as ISDN or mobile phones to the VoIP network.

The establishing and tearing down of media connections in a VoIP network is handled by the Call Control and Signaling Plane. The media gateway controller or the call agent functions in this plane.

Devices like the application server and the feature server are present in the Service and Application Plane. This plane enables the control and logic functions that can be availed on the VoIP network. The Management Plane co-ordinates with the other planes using the SNMP protocol. It enables services necessary for service provisioning and billing.

OSDL

The Open Source Development Labs (OSDL) has started the Mobile Linux Initiative in order to push the adoption of Linux-enabled mobile phones. Linus Torvalds, who developed Linux, is employed with OSDL. eweek.com reports:

Linux is already among the top three OSes in "converged" mobile phones, according to industry analysts, and has shipped in Motorola handsets since 2003.

Read More: OSDL Aims Multivendor Initiative at Linux Mobile Phones

IP PBX comparison

A comparison of large IP PBXs over here.

IPv6

VoIP networks are increasingly being used by companies to avail enhanced functionalities such as unified messaging and increased mobility and savings on international calls. VoIP is being used for managing virtual contact centers as well.

However, the growth of VoIP is limited by the absence of inbuilt QoS in the IP networks that prevents it from offering levels of service that would be acceptable to the industry. The transfer of VoIP packets over firewalls is hampered due to network address translation (NAT) and protocol considerations. This issue, along with eavesdropping concerns, is being considered in the development of the new generation of VoIP networks. These networks will be based on IPv6 and will concentrate on providing scalability and industry-level reliability. This would enable VoIP networks to achieve the end-to-end interworking any time and any place, which is not possible currently.

IPv6 aims to offer a better solution to the NAT-related problems as compared to the NAT-based accommodation, which is the currently used solution. The complexity and cost overheads of the Internet and its applications increase due to NAT techniques.

IPv6 will facilitate expanded addressing, autoconfiguration, multicast, QoS, etc. IPv6 will also enable a more efficient use of IP addresses by creating a new format for the addresses. The addresses will be of 128-bit each and there will be approximately 3.4 x 10 raised to 38 addresses.

Call quality testing

Factors affecting call quality include noise, echo, variations in the signal volume, etc. Voice quality is tested for listening quality, conversational quality, and transmission quality. Listening quality is a subjective assessment by listeners of what they hear. Delay, echo, ease of two-way communication including listening quality is rated when conversational quality is tested. Network service quality and the quality of the network connection are measured while checking the transmission quality.

The Absolute Category Rating (ACR) is a popular subjective test for testing voice quality. The test uses a scale of 1 to 5. The Mean Opinion Score (MOS) is calculated from the individual scores and should ideally be taken from a pool of at least 16 participants. MOS scores offered by companies for their codecs are subjective scores that are influenced by a number of variables. Laboratory testing of voice is done using phonetically balanced text such as the Harvard Sentences. This helps in obtaining a subject’s reaction to a voice that covers the whole range of sounds found normally in speech.

Degradation Category Rating (DCR) and Comparison Category Rating (CCR) are other examples of subjective tests. The amount of degradation that occurs with the damaged files is measured by the DCR and a DMOS score is given. Pairs of files are compared by the CCR and a CMOS score provides the results. The ITU distinguishes the scores as Subjective, Objective, and Estimated.

P.861 and P.862 are objective measurement techniques developed by the ITU. ITU developed P.861 (PSQM) and the newer P.862. Transmission systems and codecs can introduce a distortion into the system. These measurement techniques contrast a reference file with the weakened signals. The reference and the actual signals are divided into small segments and the Fourier Transform coefficients for each segment are calculated and compared.

The amount of distortion can be measured using these algorithms only if both the source file and the output files are accessible to the algorithms. It needs to be understood that this particular type of algorithm needs a high processing speed, i.e. processing capabilities for 8000 samples per second for narrowband voice and 16000 samples for wideband voice. The processing and memory capabilities required are quite high and in such cases a packet-based network is preferably used.

The E Model for testing VoIP quality was developed by the ETSI. VQmon® is a voice testing technology that requires far less processing power than the PSQM approach. The E Model is used to rate the transmission quality, denoted as “R”. It is a measure of what are commonly known as the “mouth to ear” factors of a conversation. The R-value has a nominal range of 0-120. For broadband telephony, the range is 50-110.

The E Model assumes that the impairments have an additive effect and is represented by the following equation:

R = Ro - Is - Id - Ie + A

Ro is a base factor. Is stands for the signal impairments. Id stands for the delayed impairments. Ie stands for the “equipment impairment factor”. A stands for the “advantage factor”.

An ACR is a subjective test and when it is performed on a wideband CODEC, the score may not be representative of the actual performance if the reference conditions are set for a narrowband CODEC.

VoIP testing

The performance of the network components is critical in deciding the success of a VoIP deployment. Applications that straddle the worlds of PSTN and IP communications utilize gateways to convert voice into IP packets and frequently these gateways need to function under high loads.

It is important that a reliable testing methodology be in place so that the quality of voice transmission can be tested. Tones can be used to test the continuity of the connections and the latency. A good VoIP testing technique should stress all the constituents of the VoIP gateway. It should be capable of conducting audio quality testing as well.

The testing methodology also needs to confirm to the quality standards followed by PSTN carriers. Stress testing is carried out test voice connections for degradation, for voice activity detection (VAD), and for complete stressing of the base signal algorithms that are used in the codecs. As VoIP is essentially a convergent technology in which voice may either originate or terminate in a PSTN network, its testing methodology should be in conformance with PSTN standards.

Testing the information signal tones for ringing, busy, etc is done for the user as well as the network equipment. In all likelihood, VoIP will become a high compression application and will have to face the same problems that wireless carriers over digital networks faced. The standards for verifying and detecting the tones were born as a result of these problems. The specifications ITU-T P.50 and P.59 relate to the use of artificial voice. In recommendation P.50, artificial voice is defined as "a signal that is mathematically defined and that reproduces the time and spectral characteristics of speech which significantly affect the performances of telecommunication systems."

The temporal behavior of human conversation, which includes pauses, mutual silence, etc, is described by recommendation P.59. This is important for the testing of the speech processing systems present in speaker phones, DCME, ATM systems, etc.

Real speech in a conversation includes pauses, noise, and variations in tone. Voice is a variable signal that also includes short portions of inter-syllabic voices. The standard speech coders such as CELP, RELP, etc use linear prediction algorithms in order to manage the variations in voice. The signals between the frequency samples are predicted by these algorithms; this allows them to manage a range of voice types with different accents.

In order to test the voice as well as the non-voice aspects of the coder architecture, a two-tier testing methodology that includes tone and artificial/real voice testing, is recommended.

VoIP in the enterprise

VoIP provides carriers and customers with several advantages, these include:

• Reduction in toll charges that one has to pay when running calls over PSTNs. Combining of voice and data helps to conserve bandwidth.

• Pursuing open standards enables businesses to purchase the best available solution and achieve interoperability between products from different vendors, something that is not possible with traditional PSTN solutions, with most of them being proprietary in nature.

• By turning voice into an IP application, vendors provide companies with the opportunity to make maximum use of the latest developments in the world of telecommunications.

The early VoIP vendors concentrated on developing toll-bypass solutions to allow companies to reduce communication costs. However, as most of the solutions were proprietary in nature, interconnectivity was not easy to achieve. The four main independent standards are H.323, MGCP, SIP, and H.248.

Currently, the VoIP networks are being deployed using multiple protocols and architectures. The combination of protocols depends upon the type of services that a company wishes to deploy. Given below is a brief introduction to the various VoIP protocols:

• H.248/Megaco is a protocol for defining the centralized architecture used in the creation of multimedia applications such as VoIP.

• H.323 defines the distributed architecture of packet-based multimedia communication systems.

• MGCP is used to define the centralized architecture for creating multimedia applications such as VoIP.

• RTP is used to define the transport protocol for real-time applications.

• SIP is used for defining a distributed architecture used for creating multimedia applications.

VoIP networks can be either centralized or distributed. The flexibility regarding the choice of architecture allows companies to develop networks that can strike a balance between ease of management and innovative service. MGCP and Megaco are protocols used for a centralized device referred to as the media gateway controller, which manages the switching logic. The endpoints in a centralized network do not have any native features and the network intelligence is centralized. In order to develop SIP and H.323 networks in a centralized manner, back-to-back user agents (B2BUA) or gatekeeper routed call signaling (GKRCS) is used.

The advantages of a centralized architecture include centralized management and call control. Legacy voice features can be replicated with ease. The drawbacks include limiting of VoIP services to legacy voice features.

With a distributed architecture, the network intelligence for call handling is distributed to the end-points as well. Call handling implies features such as call state, calling features, call routing, billing, etc. A VoIP call can be initiated and terminated at the VoIP gateway, media server, IP phone, etc. The devices that do the call controlling are referred to as gatekeepers and redirect servers in H.323 and SIP networks, respectively. The disadvantage of a distributed architecture lies in its complexity. Companies looking at interconnecting the various segments by using VoIP protocols can do it in three ways:

• By using TDM tools or VoIP gateways for translating between protocol domains. This particular model is viewed as a stop gap arrangement till translators that are IP-based are available. Using this protocol increases latency and adds a protocol translation to the process.

• A single protocol architecture allows the company to run all the devices on a single protocol. This allows the company to keep the network simple but limits the ability to migrate the existing applications to the new protocol and connectivity to other networks that are using different VoIP protocols may be difficult.

• By employing IP-based protocol translation, two or more VoIP protocol domains can be connected. A company can continue to use its existing equipment while using IP translators; unlike TDM connections these do not introduce a delay. However, there are no standards for IP-based translators as yet.

In conclusion, it can be said that the selection of VoIP protocols depends upon the technical and service requirements of a company. In choosing a vendor, it is safer to select one that has developed its applications on open standards so that interoperability with other VoIP systems is not a problem. The applications should support multiple protocols to facilitate addition of new products or migration to other systems, without having to perform upgrades every time. A system that supports a multi protocol environment allows a company to develop a scalable network.

VoIP QoS II

Along with intserv, another service developed by IETF to fulfill the QoS requirements is diffserv. intserv is not really a built-in service as it requires per hop signaling and reservation of resources along the route that the data is going to take. Also, intserv does not support scalability of a network that is becoming increasingly complex. Differentiated Services or diffserv do not reserve bandwidth but accomplish the transfer of information by using a field in the IP header.

The operations of diffserv are detailed in RFC 2474, RFC 2475, and RFC 3260. Essentially, diffserv is used to distinguish and prioritize between Internet services. This can be done in a variety of ways such as priority levels based on price levels, application requirements, etc; network attributes such as jitter, latency, etc. Network services are provided based on the definitions given by the diffserv architecture. This enables the provision of the available bandwidth to the traffic streams. The packets are classed into specific types for routing purposes.

The packets move from the source to the destination based upon the markings on them. diffserv utilizes the Differentiated Services (DS) field in IPv4 and the Traffic Class field in the IPv6. diffserv uses six of the eight fields that are present in the DS field. The six bits are collectively referred to as the Differentiated Services Code Point (DSCP). DSCP allows for 64 Internet service distinctions. According to IANA, IETF will standardize 32 codepoints and the remaining codepoints will be used equally for local and experimental use and for probable standardized assignments as the need occurs.

TCP/IP and VoIP

The Transmission Control Protocol (TCP) is a connection-oriented protocol. Internet Protocol (IP) and User Datagram Protocol (UDP) are connectionless protocols. In a connection-oriented network, a network has to be established before information can be transferred. In this process, a significant amount of time and resource is spent in signaling. The advantage of this process is that upon the establishment of a network path, attributes of the path such as propagation delay do not change. In fact, connection-oriented networks are also known as reliable networks.

Data networks are examples of connectionless networks in which the data travels in packets. The packets may or may not be delivered to the desired destination. Since the characteristics of the path are not fixed, a connectionless network is often described as unreliable or best-effort. File transfer using FTP, electronic mail using SMTP, and remote host computer access using Telnet were important data applications that were developed using TCP, IP, and UDP. Today, the focus is on applications that integrate voice and data. As these applications are intolerant of delay, they cannot be run on a best-effort network. The answer to the problem lies in adding protocols that will boost the performance of connectionless IP networks. Given below is a list of protocols that enable voice and data transfer over the Internet.

•Multicast Internet Protocol enables the transmission of information from a single source to many recipients.

•RTP Control Protocol checks the performance of the RTP.

•Real-time Streaming Protocol enables data delivery in real-time, which also includes accessing information from media servers.

•Real-time Transport Protocol helps in payload identification and sequence numbering.

•Resource Reservation Protocol ensures that there is sufficient bandwidth available to enable communication between sender and receiver.

•Session Announcement Protocol packets help the end users to make use of open sessions.

•Session Description Protocol allows information exchange regarding the media stream, bandwidth required, session name, etc.

VoIP standards

VoIP is a new technology and still does not have universal standards. This makes it difficult for network managers to integrate the products obtained from different vendors. In the absence of standards, vendors come out with proprietary standards that make interoperability a difficult proposition for enterprises. This is where the “standards bodies” come into the picture. These bodies are made up of the inventors, developers, vendors who have an interest in a particular technology.

The International Telecommunications Union (ITU) and the Internet Society define VoIP standards. The ITU has its headquarters in Geneva and was established in the 1860’s in order to develop standards for telegraph communications. The ITU is divided into ITU-R, ITU-T, and ITU-D. These are the Radio Communication Sector, the Telecommunications Standardization Sector, and the Telecommunications Development Sector, respectively. The ISDN and ATM standards for telecommunications have been developed by ITU-T. The standards can be identified on the basis of a letter that is assigned to a particular aspect of that technology. ITU-T standards that begin with H relate to audiovisual and multimedia. VoIP is covered under this group of standards. ITU-T standards can be viewed online. The Internet Society has been involved in issues related to the Internet since 1992. It concentrates more on packet switching and data transmission issues.

The Internet Society also works as small groups such as the Internet Architecture Board (IAB), Internet Research Task Force (IRTF), Internet Engineering Task Force (IETF), etc. Internet Standards, also known as Request for Comments or RFC documents are developed by the IETF. Some well-known RFCs include the Hypertext Transmission Protocol (HTTP), RFC 2616, and the Session Initiation Protocol (SIP), RFC 3261. Organizations such as the American National Standards Institute and the European National Standards Institute also influence standards but at a less broad level.

A company that wishes to implement VoIP should try and get an understanding of the standards that govern their VoIP devices and applications. This is because applications that follow the ITU-T specifications may have different networking and architecture issues than those that follow the IETF standards. Knowledge of the standards will help in making the right product decisions and also help to solve interoperability issues.

VoIP Telephony, The Need of the Hour

In a world which is prone to changes in technology, VoIP is seen as a major invention. The unprecedented growth of VoIP users has proved that VoIP is going to dominate the telephony market in the coming years. With the announcement of Vonage, a leading VoIP provider that it has reached one million customers is pointed at the trend for the future. Vonage is not the only company, which is expanding its VoIP network. Many other companies believe that VoIP is the need of the hour. They are confident that providing VoIP services to the customers, they can satisfy all their requirements.

It is beyond doubt that consumer VoIP is on the rise. The mainstream telephone users are now eager to avail low cost Internet telephony and the improved features which are provided by VoIP. news.ft.com reports:

Unlike most traditional phone calls, calls based on VoIP technology are digitised, chopped up into tiny electronic packets and then sent to their destination over the public internet. That translates into more efficient use of bandwidth and lower costs for VoIP service providers.

Read More: Why VoIP telephony is quickly coming of age

VoIP protocols - Part 3

SAP stands for Session Announcement Protocol. It is an announcement protocol used in the advertising of multicast media by the session directory clients. It is also used in communicating the session setup information to participants. The multicast announcement has the same scope as the session that it announces, this helps in keeping the local session announcement local and in maintaining the scalability of the protocol.

A SAP listener uses the Multicast-Scope Zone Announcement Protocol for listening to the multicast scopes on a SAP address and port. Instead of IPsec authentication headers, application level security is used in facilitate interoperability between mechanisms that are used for announcing the sessions. The session can be announced by a web page, a session initiation protocol, or by email.

The SAP protocol structure includes:

• V: A version number field, which is three bits and is set to 1.
• R: It stands for Reserved and is set to 0.
• T: It is the message type and can have a value of 0 or 1, where 0 is the session announcement packet and 1 is the session deletion packet.
• A: It is the Address Type and can have a value of 0 or 1. 0 is the originating source field and contains a 32-bit IPv4 address, 1 is the originating source and contains a 128-bit IPv6 address.
• C: It is the compressed bit and the payload is compressed if C has a value of 1.
• E: It is the encryption bit and can have a value of 0 or 1. 0 implies that the packet is not encrypted and that the timeout must be absent and 1 implies that the payload is encrypted and the timeout field has to be present at the packet header.
• Timeout: It is a value that gives the NTP time for timing out a session and is included when the session payload has been encrypted and in the absence of the decryption key, listeners may not realize the timing fields in the payload.
• Payload type: It specifies the MIME content type and it elaborates on the payload format.

SDP: It stands for Session Description Protocol and elaborates on session announcement and session invitation. A session directory tool present on the Internet Multicast Backbone (Mbone) helps in advertising the conference sessions and provides the conference address and other relevant information. The SDP messages, which are UDP packets, are relayed by multicasting an announcement packet to a popular multicast address using SAP. The messages carry a SAP header and a text payload and can be sent across the World Wide Web by using email. SDP uses different transport protocols such as SAP, SIP, RTSP, HTTP, etc. Also, SDP does not support session content negotiation and media encodings. The SDP messages consist of the session name, its duration, media details, and necessary information to access the media.

SIP: SIP stands for Session Initiation Protocol and it is an application-layer protocol that provides mechanisms for end user systems and proxy servers to establish, change, and end multimedia sessions including VoIP calls. It can also be used to initiate multicast conferences. Existing sessions can be modified by adding or removing media from it. Names can be mapped and services can be redirected with the help of SIP. This enables user mobility as a user can now have a single identifier independent of their location. SIP supports the following facets of multimedia communication:

• User Location: It helps in determining the end system to be used in the communication process.
• User Capabilities: Media parameters are determined by these.
• User Availability: It checks for the readiness of the receiver to participate in a communication.
• Session Management: It helps in modifying session parameters and ending sessions.
• Session Setup: It sets up the session parameters at the caller's and the receiver's end. 

SIP can be used as a component to develop a multimedia architecture like RTP that can be used to provide real-time data as well as feedback on QoS; the delivery of streaming media can be managed by RTSP; gateways to PSTN networks can be controlled by Megaco; and SDP can be used for providing information on the multimedia sessions. SIP can be used along with these protocols, however, its functioning is not impeded in the absence of these protocols. SIP is also used to provide security against DoS attacks, facilitate user to user and proxy to user authentication, and encryption.

In an Internet telephony session, SIP addresses are used to identify the caller and the receiver. A caller making an SIP call sends a request to the relevant server. The request may reach the receiver directly or it may lead to a number of SIP requests by the proxies. The SIP addresses can be present on web pages in the form of URLs, this helps in integrating them with applications like Click to talk.

The SIP messages can be sent using TCP and UDP, the messages are text based and use the UTF-8-encoded ISO 10646 character set. The messages are either requests or responses. The lines end with CRLF. An SIP request message consists of

• Method: Methods include Invite, Ack, Options, Bye, etc and are carried out on the resource.
• SIP version: The version of the SIP.
• Request-URI: It is the SIP URL or the general Uniform Resource Identifier to which the request is addressed.

A response message header has the following format

• Reason-phrase: It describes the status code.
• SIP version: The version of the SIP
• Status-code: It is an integer code that relates to the efforts to fulfill a request.

SGCP: It stands for Simple Gateway Control Protocol and it is an Internet protocol within a distributed system and is used to control telephony gateways, which are basically network elements that facilitate conversion between audio signals and data packets that are transferred over various networks. The SGCP works as a connection model and its two primary components are endpoints and connections. Call agents set up the connections that are grouped in calls. An endpoint consists of a domain name of a gateway and a local name inside the gateway.

VoIP protocols - Part 2

MIME stands for Multipurpose Internet Mail Extensions and is a set of standards that redefines the format of messages to accommodate character sets for message bodies. These character sets are different from US-ASCII. The headers that define the structure of MIME messages are covered under RFC 2045. The initial set of media types is defined by RFC 2046. RFC 207 elaborates on the extensions that permit non-US-ASCII text data in Internet mail header fields. IANA registration procedures are specified by RFC 2048. MIME message formats and acknowledgements are illustrated by RFC 2049.

MIME enables an email to carry almost any type of text, image, audio, and video data. MIME employs base64 as an encoding procedure to ensure protection for non-text messages. It achieves this objective by coding non-text messages as text. Communication protocols such as HTTP also use MIME for the transmission of data. Messages are mapped in and out of a MIME format by email clients.  MIME was developed under the condition that the existing email servers would not need any changing. This is made possible by making the MIME attributes optional. It is possible for a MIME-capable client to interpret a non-MIME message by using its default values.

MIME type comprises a combination of type and subtype. The charset of a text type reveals its encoding. Internet protocols such as HTTP use the content-type header and MIME type registry. MIME enables messages to have a tree structure. MIME supports the following message types:

• text messages of the text/plain type, this is the default value for "Content-type:".
• text with attachments, this is of the type multipart/mixed with text and non-text parts. The MIME content-type and the filename extension indicate the type of file.
• original attached to the reply, this is of the type multipart/mixed with the original message included as a message/rfc822 part.
• messages sent in alternative formats such as HTML in which the messages are of the type multipart/alternative and having the content in other formats like text/html.   

RVP over IP: It is a proprietary specification developed by MCK Communications. It is used for the transfer of digital telephony sessions over packet networks. The signaling occurs through the TCP session and the voice is transferred via the UDP session. RVP over IP depends on the network configuration and the level of QoS. MCK offers proprietary PBX services and RVP over IP is used for connecting a remote client and a phone switch. When a remote caller attempts to make a connection with the PBX, a TCP session to the Extender PBXgateway is initiated by the MCK Extender. The initiation occurs from a high TCP to TCP 2698.

The devices communicate as client and server with the MCK Extender products functioning as clients. The first TCP port to begin with 1024 or higher is opened by a client that initiates the RVP over IP session. The client then sends a request to TCP 2698. Voice and network parameters make up the data packets. The voice parameter consists of a voice path, voice compression algorithm, DTMF encoding, comfort noise generator, echo cancellation, silence detection. The network parameters comprise packet size and jitter buffer. The remote MCK extender starts the UDP stream upon the successful establishment of the TCP session. The UDP stream starts from port 12288 (0x3000) up to 12544 (0x30FF).  The UDP listening port is 2698. RVP over IP reduces network traffic congestion and packet loss by employing a packetizer that uses a data packet for holding several voice samples.  The codec and packet size determine the interval at which voice is transmitted.

VoIP protocols - Part 1

The growth of VoIP can be attributed to the low cost and integrating of voice and data infrastructures. The components that make up a VoIP system include a Signaling Gateway, Call Manager, Call Agent, and Media Gateway Controller. The Gateway is capable of converting the media from one type of network into the desired format. Duplex media translations, T.120, audio, and video can be processed by the Gateway, which can also perform media conferencing and play audio/video messages. The voice signal is fragmented into frames by the digital signal processor (DSP). The voice signals are transmitted as voice packets across networks such as T.38 (ITU), MGCP (level 3, Bellcore, Cisco, Nortel), SIP (IETF), H.323 (ITU), SIGTRAN (IETF), etc. The bandwidth utilization is governed by coders as per the techniques mentioned in ITU-T recommendations such as G.723.1 and G.729. RTP/UDP/IP are the protocol stacks used for the conversation, which is initiated by the enabling of the codecs at both ends of the connection. VoIP provides services such as phone to phone, PC to phone, phone to PC, fax to e-mail, e-mail to fax, fax to fax, voice to e-mail, IP Phone, transparent CCS (TCCS), toll free number (1-800), class services, call center applications, VPN, Unified Messaging, Wireless Connectivity, IN Applications using SS7, IP PABX and soft switch implementations.

QoS: Quality of Service is an important aspect of VoIP communication, it covers facets such as delay, jitter, echo, packet loss, packets arriving out of sequence, etc. The QoS of a VoIP service is determined by the Mean Opinion Score (MOS). The quality of voice varies with the CODEC and MOS is used to evaluate the quality of voice for a given CODEC. PSQM (ITU P.861), PAMS (BT), and PESQ are algorithms that have been created to measure the QoS.

Megaco: It stands for Media Gateway Control Protocol and is designated H.248. It is used for separating call control from media conversion in a physically decomposed multimedia gateway. H.248 esentially defines the relationship between a Media Gateway (MG) and the Media Gateway Controller (MGC). Circuit-switched voice is converted to packet-based traffic by MG whereas the MGC controls the service logic of the traffic. H.248 can even support ATM networks, something that is not possible with MGCP. Moreover, the signaling systems supported by these network interfaces include ISDN, ISUP, QSIG, GSM, etc. Streams of data from outside a packet can get connected to a packet or a cell stream like the RTP protocol. H.246 provides a structure for gateways and IVRs. H.246 consists of two primary components, namely terminations and contexts. Analog telephone lines and MP3 streams are examples of terminations that are either entering or exiting the MG. The MGC can alter the properties of the terminations. By adding and removing the first and last termination, contexts can be either created or released. Megaco uses several commands to manage the terminations, contexts, events, etc. These commands include the following:

• Add: It is used to create a Context when executed on the first Termination in a Context.

• AuditCapabilities: This command lists the values for the Termination that are permitted by the Media Gateway.

• Modify: The events and signals in a termination can be modified by the Modify command.

• AuditValue: The latest statistics of a Termination can be learnt with the help of this command.

• Notify: The occurrences in the Media Gateway are made known to the Media Gateway Controller by the Media Gateway with the help of this command.

• Subtract: It is used to separate a Termination and its Context, when used on the last Termination in a Context, the command deletes the Context.

• Move: A Termination can be shifted to another Context by executing this command.

• ServiceChange: The MG informs the MGC via this command that a Termination has either been removed from service or is being reintroduced into service.

Media Gateway Control Protocol (MGCP): It is a VoIP protocol that is used for controlling telephony gateways, i.e. a Call Agent and a media gateway. The Call Agent houses the call control intelligence whereas the media gateway comprises media functions such as converting TDM voice to VoIP. It is between the Call Agent and the media gateway that the audio signals and other data packets that travel over the Internet undergo conversion.

A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks. Thus, it can be said that the MGCP is a master/slave protocol where the Call Agents give the commands and the gateways execute them. The MGCP is basically a connection type of model with the endpoints and the connections being the basic components. The endpoints can be both physical and virtual. There are two types of connections, point to point and multipoint. Data can be transmitted between two endpoints by establishing a point to point connection between them. A multipoint connection can be established by connecting the endpoints to a multiple session. In an MGCP model, the signaling layers of the H.323 standard are implemented by the Call Agent, which presents itself as a Gatekeeper. Transactions in an MGCP model consist of a command and a mandatory response. There are eight types of commands in MGCP, these are:

• CreateConnection: It defines the receive capabilities of endpoints by using SDP.

• ModifyConnection: Similar to CreateConnection, it modifies the properties of a connection.

• AuditEndpoint: It reveals the status of an endpoint.

• Notify: The media gateway controller is informed upon the incidence of an event.

• DeleteConnection: It ends a connection and provides relevant statistics about the connection.

• NotificationRequest: The media gateway receives requests to despatch notifications on the incidence of particular events in an endpoint.

• AuditConnection: It details the parameters relevant to a connection.

• RestartinProgress: It is used to indicate whether an endpoint is either in or out of service.

H.323 protocols and their applications - Part 3

H.261: It is basically a video coding standard that is used for transporting via the RTP with any of the protocols that support RTP. H.261 supports CIF and QCIF video frames that have luma resolutions of 176 x 144 and 88 x 72, respectively. An H.261 header consists of SBIT, EBIT, I, V, GOBN, MBAP, QUANT, HMVD, VMVD.

H.263: It defines the payload format for the H.263 bitstream in RTP. The RTP packet can use any of the three modes for the payload header of H.263. Fragmentation at the Group of Block (GOB) boundaries is supported by mode A, which is the shortest payload header. Fragmentation at the Macroblock (MB) boundaries is supported by the long payload headers, i.e. modes B and C. The H.263 payload header, whose size depends upon the modes, succeeds the RTP fixed header and the H.263 header is followed by the H.263 compressed bitstream. Payload header size is 4, 8, and 12 bytes for modes A, B, and C, respectively.

RAS: It stands for Registration, Admission and Status (RAS). It is a protocol that is used to carry out signaling functions such as registration, admission, disengagement actions, etc between the endpoints and the gatekeeper. A Request in Progress (RIP) message is used by an endpoint or a gatekeeper that is unable to respond to a request inside the timeout limits. This allows the receiving endpoint/gatekeeper to reset its timeout timer. Timeouts and retrys are particularly significant as the reliability of the RAS message channel is not very high. Important RAS messages include RegistrationRequest (RRQ), AdmissionRequest (ARQ), BandwidthRequest (BRQ), DisengageRequest (DRQ), InfoRequest (IR), etc.

RTCP: It stands for RTP control protocol and it monitors the QoS of an IPv6 RTP connection. It depends upon the multiplexing of data and the control packets. It works together with RTP in the delivery of multimedia data and provides an out-of-band control information. The RTCP header consists of the Version, P (Padding), Reception report Count, Length, etc.

RTP: It is a standardized packet format for the transmission of audio and video over multicast and unicast network services. RTP helps in payload-type identification, delivery monitoring, time stamping, and sequence numbering. The data transfer over RTP is facilitated by RTCP that enables the overseeing of data in such a manner so as to facilitate the same over multicast networks. RTP does not cover address resource reservation. RTP and RTCP function independently of the transport and network layers.

T.38: It is an Internet Fax Protocol. It deals with the transfer of fax documents in real-time over an IP network by using either TCP or UDP as per the service environment. The TCP/UDP payload carries the T.38 data. T.38 can manage fax as well as voice data over a single network. T.38 allows the use of H.323 in the same way as it is used in VoIP.

T.120: It is a family of protocols that cover the services for multilayer protocols, MCU, etc. It promotes greater operating powers that are not possible with H.231 and H.243. The Multipoint Communication Service Protocol (MCS) is covered under T.125. The procedures included in this protocol include the exchange of MCS data between two parallel MCS providers, exchange of MCS primitives between MCS providers and users. A single MAP connection or one or more transport connection make up an MCS connection.

H.323 protocols and their applications - Part 1

H.323 is a protocol that covers the broadcast of real-time voice, video, and data over IP-based networks. H.323 is applicable to multipoint-multimedia transmissions and provides a range of services that find use in various businesses. The streams of media move along RTP/RTCP, where RTP is the carrier for the media and RTCP carries the status. H.323 is an umbrella protocol from the International Telecommunications Union (ITU). H.323 also ensures the congruence of mobile multimedia applications and different services. It comprises several protocols as explained below.

DVB: Digital Video Broadcasting (DVB) systems are supported by CATV infrastructures. DVB uses satellite, cable, and terrestrial means for the purpose of broadcasting. DVB standards were created in 1993 in Europe with the objective of unifying the framework for all delivery systems.

H.225: It is a standard used to establish a call over a Registration and it encompasses narrow-band visual telephone services as per the recommendations of the H.200/AV.120-Series. H.225 deals with managing audio and video information on a packet based network in order to facilitate services in an H.323 environment. 

H.225 Annex G:  It supplements the H.225.0 RAS protocol in fulfilling the needs of communication between administrative domains. It details the process of address resolution and access authorization necessary for completing calls between administrative domains. The H.225 Annex G is required because of the amount of equipment that exists in an H.323 network. It does not require a particular system architecture inside the administrative domain and it supports different call models like gatekeeper routed and direct endpoint.

H.225E: It deals with the implementation of UDP and TCP based protocols by using a packetization method, a signaling framework, and wire-protocol. It also specifies the profile for transmitting H.225.0 messages. It uses the security measures available under IP-SEC and H.235 and its design facilitates its use in engineering networks.

H.235: It is a security protocol for the H.3xx series. It provides authentication and integration for H.323 based systems, it enables the identification of an individual and not the application. H.235 messages are encrypted in the same way as those in ASN.1. H.235 provides point-to-point and multipoint conferencing for all terminals where H.245 is used as a control protocol.

H.323(SET): It elaborates on the standards for Simple Endpoint Types (SET). SET devices are meant for a single purpose only, they constitute a large number of H.323 capable end systems. These devices enable audio calls with other H.323 endpoints while using only a small portion of the H.323 specifications. SETs need not necessarily be PC-based, they can be relatively inexpensive applications such as the telephone.

H.245: It describes the line transmission of non-telephone signals. The features described include the sending and receiving  properties and the desired modes at the receiving end as well as Control and Indication. H.245 messages can be divided into request, response, command, and indication messages. The message sets include Terminal capability messages, Logical channel signaling messages, Request Mode messages, Round Trip Delay messages, etc.

Know your wireless standards

802.11a, 802.11b, 802.11g, and Bluetooth are wireless standards that SOHOs and SMBs should know about as these may influence the type of WLAN equipment they purchase and the services that their vendor provides. 802.11 was the first WLAN standard that was created by IEEE in 1997, it supported a bandwidth of 2 Mbps only. It was followed be 802.11b in 1999. 802.11b supports a bandwidth of 11 Mbps and uses a radio frequency of 2.4 GHz, which is an unregulated frequency and hence affected by interference and disturbance from applications like ovens and cordless phones. 802.11b is very low cost and its signals have a good range and suffer from minimum obstruction. However, it is slow and cannot support multiple users at the same time. 802.11a was created at around the same time as 802.11b and caters to the business market. It can work with a bandwidth of up to 54 Mbps and regulated signals of up to 5 GHz. The higher frequency reduces the range of 802.11b and the signals cannot penetrate obstructions easily. As the two standards work on different frequencies, they are not mutually compatible but can be implemented alongside each other. 802.11a provides a very high maximum speed and multiple users can work on it simultaneously. 802.11g is a combination of 802.11a and 802.11b. It supports a bandwidth of 54 Mbps and a frequency of 2.4 GHz. 802.11g network adapters are compatible with 802.11b network adapters. It offers the advantage of a very high maximum speed and a high signal range. An alternative to WLAN is Bluetooth, which has the advantage of being a low-cost technology but is unable to support a high bandwidth. It works with a bandwidth of 1 Mbps and at a range of 10 meters. Therefore, even though it can network cell phones and PCs, it does not offer much value for WLAN networking. 

Senator Sunuhu Calls for National VoIP Standards

Senator John Sunuhu (R-NH) addressed a VON conference and described what he hoped would become a major telecom bill in 2005.

According to eWeek:

Sununu said that next year he and others in a Senate committee, under the leadership of Sen. Ted Stevens (R-Alaska), will be writing a broad telecom act to deal with broadband voice, universal service reform and some spectrum issues. Universal service fees are those subsidies paid to carriers in areas of low population density, to compensate them for lower subscriber revenue.

The "Regulatory Freedom Act" (S2281) that Sununu submitted to the Senate Commerce Committee in July passed the committee hearing "with one problematic amendment that gave some power back to state regulators to impose [universal] service fees at the state level," Sununu said.

Read more: Senator at VON Calls for National VOIP Standards