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Wednesday, February 7, 2007
How many times has it happened that you've received a nagging call on your cellphone by someone trying to sell you a credit card just when you're in the middle of a crucial meeting, or maybe enjoying an afternoon nap on a holiday? In another case, how many times have you wished that a person had called you instead of seeking clarification about something via email? While you were available on the phone, the sender didn't know and out of the fear of disturbing you, just shot you a mail instead. Let's take the reverse case. How many times did you try to reach a colleague for some urgent information, but hit a stone wall? You first called on the office extension number, but the person wasn't there on the seat. You then called on the person's mobile, which went to voice mail. You then sent a sms, but didn't get a reply back. Finally, as a last resort, you just sent an email asking the person to contact you when free. Later, the person read your mail and tried to contact you, but faced similar obstacles. Finally, after wasting so much time, effort, and money on phone calls, you both just ended up discussing things over email. All these situations are not a new phenomenon.
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Concepts, building blocks & implementations
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Chances are that you face them everyday. In fact, we all face them everyday, which is quite sad because despite having so many different modes of communication, we're not able to reach the people we want to when we really need to. On an average, every corporate employee today uses at least three to four modes of communication-landline, mobile, email, and chat. And yet, reaching people is becoming increasingly difficult. Not being able to reach a person is not only frustrating, but also counter productive and expensive with so many modes of communication. It can also lead to delays in key decisions being taken, which can lead to other problems. What's needed is an integrated solution that will tell you whether a person is available and the best way of reaching him/her. That's essentially what unified communications proposes to do.
A successful unified communication deployment is one where you're always contacted through the mode you prefer, and by the people you want to be contacted.
While all this may sound like a distant dream, it actually isn't. In fact, the technologies have already arrived, and the products and solutions are also available. It's just a matter of time before we all start taking it for granted, like email and mobile phones. In this story, we'll tell you how close you really are to unified communication, and even do some sample deployments. Though not full-fledged, the deployments will give you an idea of the possibilities. The rest is up to you to explore.
Basic building blocks
A typical UC solution is a well-knit integration of various elements like e-mail, chat, telephones (PBX or GSM), collaboration tools, audio and video-that form the basic building blocks of such a system. Of course, you cannot forget IP, which forms the backbone of this integrated system. This is because you need a standard medium so that all these building blocks speak the same language and, therefore, communicate with each other. As a result, PBX becomes IP PBX, audio goes to become VoIP and enables audio conferencing over IP, e-mail takes up more messaging and collaboration features, there is web conferencing and unified messaging.
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Note that while IMs provide almost all the functionalities from mails, voice chat, PC-to-PC calling and connectivity over SMS-all this is currently happening in the public domain. So all you need now is to implement these functionalities at the corporate level. Now let us see the role of each of these blocks separately to make an efficient UC system.
IP PBX- An IP based PBX system allows an organization to converge both voice and data networks, over a LAN or a WAN. It uses standard packet-switch protocols to carry voice across a data network. There are state-of-the-art IP-based solutions that allow you to integrate incoming as well as outgoing voice applications with applications that run on the Net like real-time chat, Web collaboration, and e-mail. As a result, you can have multiple interactions through a single agent simultaneously, no matter which communication channel is being used by the customer.
VoIP- As you're using the same network for voice and data communication, you save on infrastructure costs. Plus, of course you can use your existing data links between your various branch offices for voice as well, thereby saving on STD bills. There are various kinds of soft and hard phones that enable VoIP as a preferred means of communication.
Presence server- is an important element that sits in the UC network and keeps a record of whether the receiver is present or not. It also records the information about the mode of communication that is being used by the user, such as whether he is available on e-mail, IM, fixed line, cellphone, etc. This information is collated and used by other call/phone manager systems or communicators that accordingly send the message to the receiver over the communication device being used by the user at that time.
E-mail-is one of the most preferred methods of written communications in organizations. But over a period of time when it has been in use, it is now time to see what more can you do with it. Today solutions such as Exchange 2007 IBM Domino have enabled user to do much more than just sending or receiving e-mail. You can do messaging and conferencing, call (telephone, fax) management, document sharing and much more on these.
Unified messaging-This is not the same as unified communication as it is very often misunderstood. Unified messaging is a subset of the larger umbrella of solutions that unified communication provides. UM keeps your communication limited to the extent of messaging and collaboration, and lets you do voice mail, e-mail, text to speech, and view fax messages from one common interface. For example, you can do all this with Outlook 2007. This is far less than the capabilities of a UC system that takes you beyond UM to add more features like collaboration/multimedia conferencing.
Audio conferencing-While this is the most commonly used mode of conferencing even today, it has added a lot of features in it to become an important element of UC. There are many solutions available in this space. Skype and public IMs are in popular, there are many commercial solutions like those from Polycom and Avaya.
Web conferencing-This is also an important component of a UC solution. There are both hosted solutions like Webex and that from Citrix, and you can also have one set up on your own over the Internet. Since everything is going over the Internet while making a UC system work, Web conferencing becomes the hot favorite as it can let you do all forms of data, voice and video conferencing.
Voice mail-This refers to sending a recorded message. Though this has been in use since long time, the real use of voice mail
beyond telephone's answering machines is being made by the UC systems. This happens when voice mail too shifts to IP, to help deliver voice mail on any network (GSM or PSTN) you are in. With a good UC solution in place, you can call into your mail server from either device to have your voice messages read out.
UC Client-Now this is pretty obvious. Unless there is one client that brings all the above mentioned components together into one console, there can be no unified communication.
There are some such products/solutions already available in the market like the one's powering MS Outlook and Office Communication Server 2007, that help one element talk to the other and respond in the manner most suitable to the user.
Integrating the blocks
To achieve the desired scenario described in the introduction, our typical UC user connects to the UC system via one of different available means: over the IP network (that can be his LAN or through the Internet), over a PSTN telephone link or from a PDA or mobile phone using a GSM or CDMA network. Therefore, the first point of integration is at the network level, where each of methods of transmissions must be brought to a common IP layer that our UC system works with. This will require an IPPBX gateway to bring the PBX (telephony) to an IP platform. Additionally to support mobile networks you will need a device that can link GSM/CDMA to IP. The next hand off can be done with software, where you can setup 'preference registries', that contain information about how different users prefer to handle their calls depending on their status. For instance, if a user sets his status to 'In a meeting', all incoming calls except from the user's boss can be sent as emails or be directed to a colleague. After this comes all the different hardware and software that let you actually do e-mail, chat, collaboration, VoIP and Video on IP. Even here, you will easily find solutions that integrate mail, chat and collaboration. To add voice and video conferencing, you may need to add one more step to the checklist.
Vendors offer unified communications in the form of different 'editions' or 'packages' that parcel different sets of components. For instance, one edition of a UC solution can have IP telephony, messaging (email and chat) and basic conferencing. A step up from there can include video conferencing and voice-based access to messages. Solutions are also available that perform the role of integrating all modes of communication but leave out their implementation to other solutions---for instance, it can let users connect to it to access their email, voice mail, faxes, phone calls and so on, but the actual IPPBX part with call routing will need a dedicated call management solution.
| Cisco Demos Unified Communications |
| The PCQuest Team visited Cisco at their Delhi office and got to experience a demonstration of Unified Communications in action.
The Cisco team demonstrated both IP phones as well as softphones in action, with the use of a Presence Server. The Presence Server was used to maintain and manage availability status of different users who were logged on from the IP and softphone terminals. Then, calls were made between all combinations of IP phones and softphones (viz. IP phone to IP phone, softphone to softphone, IP phone to softphone and so on). After that, we were also shown how the IP phone call will always follow the user, regardless of which terminal he is logged on from and even if he switches from hardware to software phones. We also saw the Cisco Call Manager in action, which is used to create rules for routing different kinds of calls and monitor them in action.
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Protocols
Although TCP/IP is the basic Internet protocol there are a few more necessary for an IP-based communication. Let's take a look.
SIP: Session Initiation Protocol is an ASCII-based standard specified in IETF RFC 2543, used to initiate, modify and terminate interactive user sessions involving multimedia. You can invite users to conferencing with multicast sessions. Since this protocol supports mapping and redirection services, you can communicate from any location. The requests from clients identified by SIP URLs, can be sent through transport protocols such as UDP, SCTP or TCP. SIP determines the end system, the communication media and partners and the called party's desire to engage in communication. Once all this is confirmed, the protocol fixes call parameters at either end with call transfer and termination.
H.323: is a binary ITU standard which ensures compatibility in videoconferencing, including call control and management, termination for both point-to-point and multipoint conferences, and gateway administration of media traffic, bandwidth and user participation. Both H.323 and SIP are 'intelligent endpoint protocols,' that locates remote endpoints to establish media streams between local and remote devices.
MGCP: Media Gateway Control Protocol describes the method of communication between a media gateway, which converts data from the circuit-switched network format to that required for a packet-switched network and the media gateway controller. It keeps track of the number of carriers and other IP telephony users.
Megaco or H.248: Listed by IETF as Megaco (RFC 3015) and by ITU-T as Recommendation H.248, Megaco/ H.248 is a call-control protocol between a gateway controller and a gateway. While MGCP makes it possible for the controller to determine the location of each communication end-point and its media capabilities, for choosing a type of service, Megaco/H.248 enhances this functionality by supporting more ports per gateway as well as multiple gateways. Other than H.323/SIP and H.248/MGCP, there are proprietary protocols from various companies such as Skype that have been quite popular. 'Skinny' or Skinny Client Control Protocol (SCCP) is a proprietary protocol used by Cisco for communications between its Call Managers and VoIP phones.
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Sreedhar Sambukumar
(Ex-GM, Software Development)
Siemens Public Communication
Networks |
| In my opinion, Unified Communication has not taken off in a big way in India. One of the challenges of a UC solution is that it doesn't necessarily adapt very well to already invested infrastructure. Many a UC solution expects a green field kind of setup for optimal benefit. Eg, if you have a classical TDM based PBX, most solutions expect digital/VoIP based standards.In India, the merger of enterprise networks with PSTN is limited due to Government regulations. |
RTP and RTCP: are two of the more common speech transmission protocols. The Real-Time Transport Protocol or RTP defines the way for programs to manage real-time transmission of multimedia data over multicast network services. It is designed to support video conferences with multiple, geographically dispersed participants. RTP combines data transport with a control protocol (RTCP), making it possible to monitor data delivery for large multicast networks. This allows the receiver to detect if there is any packet loss and to compensate for any delay jitter. Both the SIP and H.323 use RTP. To ensure security, RTP was improved to Secure RTP (SRTP), defined in RFC 3711. SRTP provides for encryption, authentication, and integrity of the audio and video packets transmitted between devices.
H.264 video codec : This video codec is used as a low bit rate compressed encoding solution for moving pictures in video conferencing. It can be used with both H.323 and SIP based solutions. It provides an umbrella for low bandwidth conferencing over ISDN and IP. H.264 enables videoconferences to connect at half the bandwidth and retain the same quality. This codec supports 60 fields per second and an advanced method of reducing the pixelation when there are a lot of motion or scene changes. H.264 eliminates the tiling seen on videoconferences, whenever there is a 'scene' change or a lot of motion.
Quality of Service
On the Internet, you need the desired bandwidth for your call else you face problems such as dropped calls, jitter, latency and poor quality reception. QoS policies ensure that transmission rates, error rates, and other characteristics can be measured and improved upon to avoid such irksome issues. They guarantee a certain level of performance depending on application request. These guarantees are important, especially in case of multimedia streaming like video conferencing, which is delay sensitive. QoS is controlled at the router level, so you need a router that lets you configure QoS for your WAN. Protocols such as Resource Reservation Protocol (RSVP) provide differing level of service for packets passing through a gateway host based on policy and reservation criteria. Data, audio and video all have different requirements on timeliness and quality. Using ATM, which lets you pre-select a level of quality in terms of service, QoS can be measured and guaranteed in terms of the average delay at a gateway, the variation in delay, and the transmission error rate.
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