Understanding SIP Architecture

SIP was designed to initiate interactive sessions on an IP network. Programs that provide real-time communication between participants can use SIP to set up, modify, and terminate a connection between two or more computers, allowing them to interact and exchange data. The programs that can use SIP include instant messaging, voice over IP (VoIP), video teleconferencing, virtual reality, multiplayer games, and other applications that employ single-media or multimedia. SIP doesn’t provide all the functions that enable these programs to communicate, but it is an important component that facilitates communication between two or more endpoints.

You could compare SIP to a telephone switchboard operator, who uses other technology to connect you to another party, set up conference calls or other operations on your behalf, and disconnect you when you’re done. SIP is a type of signaling protocol that is responsible for sending commands to start and stop transmissions or other operations used by a program. The commands sent between computers are codes that do such things as open a connection to make a phone call over the Internet or disconnect that call later on. SIP supports additional functions, such as call waiting, call transfer, and conference calling, by sending out the necessary signals to enable and disable these functions. Just as the telephone operator isn’t concerned with how communication occurs, SIP works with a number of components and can run on top of several different transport protocols to transfer media between the participants.

Overview of SIP

One of the major reasons that SIP is necessary is found in the nature of programs that involve messaging, voice communication, and exchange of other media. The people who use these programs may change locations and use different computers, have several usernames or accounts, or communicate using a combination of voice, text, or other media (requiring different protocols). This creates a situation that’s similar to trying to mail a letter to someone who has several aliases, speaks different languages, and could change addresses at any particular moment.

SIP works with various network components to identify and locate these endpoints. Information is passed through proxy servers, which are used to register and route requests to the user’s location, invite another user(s) into a session, and make other requests to connect these endpoints. Because there are a number of different protocols available that may be used to transfer voice, text, or other media, SIP runs on top of other protocols that transport data and perform other functions. By working with other components of the network, data can be exchanged between these user agents regardless of where they are at any given point.

It is the simplicity of SIP that makes it so versatile. SIP is an ASCII- or text-based protocol, similar to HTTP or SMTP, which makes it more lightweight and flexible than other signaling protocols (such as H.323). Like HTTP and SMTP, SIP is a request-response protocol, meaning that it makes a request of a server, and awaits a response. Once it has established a session, other protocols handle such tasks as negotiating the type of media to be exchanged, and transporting it between the endpoints. The reusing of existing protocols and their functions means that fewer resources are used, and minimizes the complexity of SIP. By keeping the functionality of SIP simple, it allows SIP to work with a wider variety of applications.

The similarities to HTTP and SMTP are no accident. SIP was modeled after these text-based protocols, which work in conjunction with other protocols to perform specific tasks. SIP is also similar to these other protocols in that it uses Universal Resource Identifiers (URIs) for identifying users. A URI identifies resources on the Internet, just as a Uniform Resource Locator (URL) is used to identify Web sites. The URI used by SIP incorporates a phone number or name, such as SIP:user@centrexorpbx.com, which makes reading SIP addresses easier. Rather than reinventing the wheel, the development of SIP incorporated familiar aspects of existing protocols that have long been used on IP networks. The modular design allows SIP to be easily incorporated into Internet and network applications, and its similarities to other protocols make it easier to use.

RFC 2543 / RFC 3261

The Session Initiation Protocol is a standard that was developed by the Internet Engineering Task Force (IETF). The IETF is a body of network designers, researchers, and vendors that are members of the Internet Society Architecture Board for the purpose of developing Internet communication standards. The standards they create are important because they establish consistent methods and functionality. Unlike proprietary technology, which may or may not work outside of a specific program, standardization allows a protocol or other technology to function the same way in any application or environment. In other words, because SIP is a standard, it can work on any system, regardless of the communication program, operating system, or infrastructure of the IP network.

The way that IETF develops a standard is through recommendations for rules that are made through Request for Comments (RFCs). The RFC starts as a draft that is examined by members of a Working Group, and during the review process, it is developed into a finalized document. The first proposed standard for SIP was produced in 1999 as RFC 2543, but in 2002, the standard was further defined in RFC 3261. Additional documents outlining extensions and specific issues related to the SIP standard have also been released, which make RFC 2543 obsolete and update RFC 3261. The reason for these changes is that as technology changes, the development of SIP also evolves. The IETF continues developing SIP and its extensions as new products are introduced and its applications expand.

Tip

Reviewing RFCs can provide you with additional insight and information, answering specific questions you may have about SIP. The RFCs related to SIP can be reviewed by visiting the IETF Web site at www.ietf.org. Additional materials related to the Session Initiation Protocol Working Group also can be found atwww.softarmor.com/sipwg/.

SIP and Mbone

Although RFC 2543 and RFC 3261 define SIP as a protocol for setting up, managing, and tearing down sessions, the original version of SIP had no mechanism for tearing down sessions and was designed for the Multicast Backbone (Mbone). Mbone originated as a method of broadcasting audio and video over the Internet. The Mbone is a broadcast channel that is overlaid on the Internet, and allowed a method of providing Internet broadcasts of things like IETF meetings, space shuttle launches, live concerts, and other meetings, seminars, and events. The ability to communicate with several hosts simultaneously needed a way of inviting users into sessions; the Session Invitation Protocol (as it was originally called) was developed in 1996.

The Session Invitation Protocol was a precursor to SIP that was defined by the IETF MMUSIC Working group, and a primitive version of the Session Initiation Protocol used today. However, as VoIP and other methods of communications became more popular, SIP evolved into the Session Initiation Protocol. With added features like the ability to tear down a session, it was a still more lightweight than more complex protocols like H.323. In 1999, the Session Initiation Protocol was defined as RFC 2543, and has become a vital part of multimedia applications used today.

OSI

In designing the SIP standard, the IETF mapped the protocol to the OSI (Open Systems Interconnect) reference model. The OSI reference model is used to associate protocols to different layers, showing their function in transferring and receiving data across a network, and their relation to other existing protocols. A protocol at one layer uses only the functions of the layer below it, while exporting the information it processes to the layer above it. It is a conceptual model that originated to promote interoperability, so that a protocol or element of a network developed by one vendor would work with others.

As seen in Figure 1, the OSI model contains seven layers: Application, Presentation, Session, Transport, Network, Data Link, and Physical. As seen in this figure, network communication starts at the Application layer and works its way down through the layers step by step to the Physical layer. The information then passes along the cable to the receiving computer, which starts the information at the Physical layer. From there it steps back up the OSI layers to the Application layer where the receiving computer finalizes the processing and sends back an acknowledgement if needed. Then the whole process starts over.

 

Figure 1: In the OSI Reference Model, Data is Transmitted down through the Layers, across the Medium, and Back up through the Layers

The layers of the OSI reference model have different functions that are necessary in transferring data across a network, and mapping protocols to these layers make it easier to understand how they interrelate to the network as a whole. Table 1 shows the seven layers of the OSI model, and briefly explains their functions.

Table 1: Layers of the OSI Model 

Layer	Description
7: Application	The Application layer is used to identify communication partners, facilitate authentication (if necessary), and allows a program to communicate with lower layer protocols, so that in turn it can communicate across the network. Protocols that map to this layer include SIP, HTTP, and SMTP.
6: Presentation	The Presentation layer converts data from one format to another, such as converting a stream of text into a pop-up window, and handles encoding and encryption.
5: Session	The Session layer is responsible for coordinating sessions and connections.
4: Transport	The Transport layer is used to transparently transfer data between computers. Protocols that map to this layer include TCP, UDP, and RTP.
3: Network	The Network Layer is used to route and forward data so that it goes to the proper destination. The most common protocol that maps to this layer is IP.
2: Data Link	The Data Link layer is used to provide error correction that may occur at the physical level, and provide physical addressing through the use of MAC addresses that are hard-coded into network cards.
1: Physical	The Physical layer defines electrical and physical specifications of network devices, and provides the means of allowing hardware to send and receive data on a particular type of media. At this level, data is passed as a bit stream across the network.

SIP and the Application Layer

Because SIP is the Session Initiation Protocol, and its purpose is to establish, modify, and terminate sessions, it would seem at face-value that this protocol maps to the Session layer of the OSI reference model. However, it is important to remember that the protocols at each layer interact only with the layers above and below it. Programs directly access the functions and supported features available through SIP, disassociating it from this layer. SIP is used to invite a user into an interactive session, and can also invite additional participants into existing sessions, such as conference calls or chats. It allows media to be added to or removed from a session, provides the ability to identify and locate a user, and also supports name mapping, redirection, and other services. When comparing these features to the OSI model, it becomes apparent that SIP is actually an Application-layer protocol.

The Application layer is used to identify communication partners, facilitate authentication (if necessary), and allows a program to communicate with lower layer protocols, so that in turn it can communicate across the network. In the case of SIP, it is setting up, maintaining, and ending interactive sessions, and providing a method of locating and inviting participants into these sessions. The software being used communicates through SIP, which passes the data down to lower layer protocols and sends it across the network.

SIP Architecture

As the Internet became more popular in the 1990s, network programs that allowed communication with other Internet users also became more common. Over the years, a need was seen for a standard protocol that could allow participants in a chat, videoconference, interactive gaming, or other media to initiate user sessions with one another. In other words, a standard set of rules and services was needed that defined how computers would connect to one another so that they could share media and communicate. The Session Initiation Protocol (SIP) was developed to set up, maintain, and tear down these sessions between computers.

By working in conjunction with a variety of other protocols and specialized servers, SIP provides a number of important functions that are necessary in allowing communications between participants. SIP provides methods of sharing the location and availability of users and explains the capabilities of the software or device being used. SIP then makes it possible to set up and manage the session between the parties. Without these tasks being performed, communication over a large network like the Internet would be impossible. It would be like a message in a bottle being thrown in the ocean; you would have no way of knowing how to reach someone directly or whether the person even could receive the message.

Beyond communicating with voice and video, SIP has also been extended to support instant messaging and is becoming a popular choice that’s incorporated in many of the instant messaging applications being produced. This extension, called SIMPLE, provides the means of setting up a session in much the same way as SIP. SIMPLE also provides information on the status of users, showing whether they are online, busy, or in some other state of presence. Because SIP is being used in these various methods of communications, it has become a widely used and important component of today’s communications.

Frequently Asked Questions | H.323 Architecture

Q: I’ve never heard of H.323. What applications do I use that rely on this?

A: Microsoft Netmeeting for one. Polycom and Tandberg videoconferencing clients are another.

Q: Do H.323 terminals have to explicitly send the H.225 call setup messages to the IP address of the gateway?

A: Yes, an H.323 endpoint must know the transport address—for example, the IP address and port number—for the Q.931 dialogue. Q.931 then provides the transport address for the H.245 control channel. This is how addresses are bootstrapped in H.323.

Q: In what layer of ISO you can put H.323 standard?

A: H.323 doesn’t map to just one layer, but is primarily implemented at layers 3 and 4.

Q: I’ve heard that H.323 uses more than one TCP/UDP port in order to transmit voice, video, and data. Are these ports fixed, or do they vary for each connection?

A: H.323 uses several ports and both TCP and UDP to signal and transport voice. H.225/Q.931 and H.245 use TCP and H.225/RAS and RTP/RTCP use UDP. Ports 1718–1720 are dedicated to H.323 traffic.

Q: Several dynamic port combinations are used per session as well.What is the best VoIP codec?

A: There are a number of factors to make that kind of determination. Probably most important is the nature of the network between the two ends. If you are connected of a LAN (high bandwidth, minimal delays, etc.), then G.711 generally provides the best voice quality.

Q: What’s an Application Layer Gateway?

A: ALGs peer more deeply into the packet than packet filtering firewalls but normally do not scan the entire payload. Unlike packet filtering or stateful inspection firewalls, ALGs do not route packets; rather the ALG accepts a connection on one network interface and establishes the cognate connection on another network interface. An ALG provides intermediary services for hosts that reside on different networks, while maintaining complete details of the TCP connection state and sequencing.

Q: What’s better, H.323 or SIP?What’s better, an apple or an orange?

A: Seriously, H.323 is based on SS7 and was designed to internetwork efficiently with the PSTN. SIP is based on HTTP and was not designed with interconnecting to the PSTN in mind. So, major carriers tend to use H.323 because it translates ISDN and SS7 signaling to H.323 VoIP signaling easily. SIP does not. On the other hand, SIP supports IM, is text-based, and is implemented more cheaply than H.323.

Exchange Unified Messaging Architecture

The Unified Messaging server role includes connections to different components, such as the Client Access or Mailbox server roles, and also to IP PBX or IP gateways, as shown in Figure 1.

 

Figure 1: Unified Messaging architecture

Generally, the UM server role communicates to an IP PBX or to a PBX using an IP gateway with the Voice over IP protocols (VoIP), Session Initiation Protocol (SIP), and Real-time Transport Protocol (RTP).

The UM server role uses MAPI protocol to communicate with Client Access and Mailbox server roles, and SMTP protocol to send voice mail messages to the destination mailbox viathe Hub Transport server. For Outlook Voice Access the UM server role accesses the mailbox using MAPI protocol to have full access to all items in the mailbox such as messages or contacts.

The Unified Messaging role no longer supports an inbound fax like Exchange 2007 UM. However, UM retains fax configuration properties, and continues to be sensitive to fax tones on calls that it answers and forwards these calls to a partner fax solution. The received fax messages look essentially the same as those created by Exchange 2007 UM, and will appear as a fax when the user is UM-enabled.

The communication to the other Exchange roles—namely the Hub Transport, the Mailbox, and Client Access Server roles—uses MAPI connections to perform tasks such as opening a mailbox for OVA or sending a voice mail message when the call has ended.

WLL Architectures | Wireless Local Loop

WLL systems come in several architectures: a PSTN-based Direct Connect network, a Mobile Telephone Switching Office/Mobile Switching Center (MTSO/MSC)-based network, and proprietary networks.

PSTN-Based Direct Connect

There are several key components of the PSTN Direct Connect network:

§  The PSTN-to-Radio Interconnect system, which provides the concentration interface between the WLL and the wireline network

§  The System Controller (SC), which provides radio channel control functions and serves as a performance monitoring concentration point for all cell sites

§  The Base Transceiver Station (BTS), which is the cell site equipment that performs the radio transmit and receive functions

§  The Fixed Wireless Terminal (FWT), which is a fixed radio telephone unit that interfaces to a standard telephone set acting as the transmitter and receiver between the telephone and the base station

§  The Operations and Maintenance Center (OMC), which is responsible for the daily management of the radio network and provides the database and statistics for network management and planning

MTSO/MSC

An MTSO/MSC-based network contains virtually the same components of the PSTN Direct Connect network, except that the MTSO/MSC replaces the PSTN-to-Radio Interconnect system.

The key components of an MTSO/MSC-based network are:

§  Mobile Telephone Switching Office/Mobile Switching Center (MTSO/MSC), which performs the billing and database functions and provides a E1 or T1 interface to the PSTN

§  Cell Site equipment including the Base Transceiver Station (BTS)

§  Fixed Wireless Terminal (FWT)

§  Operations and Maintenance Center (OMC)

For digital systems such as GSM and CDMA, the radio control function is performed at the Base Station Controller (BSC) for GSM or the Centralized Base Site Controller (CBSC) for CDMA.

In GSM systems, there is a Base Station System Controller (BSSC), which includes the Base Station Controller (BSC) and the transcoder. The BSC manages a group of BTSs, acts as the digital processing interface between the BTSs and the MTSO/MSC, and performs GSM-defined call processing.

In CDMA systems, there is a Centralized Base Site Controller (CBSC), which consists of the Mobility Manager (MM) and the transcoder subsystems. The MM provides both mobile and fixed call processing control and performance monitoring for all cell sites as well as subscriber data to the switch.

As in PSTN-based networks, the FWT in MTSO/MSC-based networks is a fixed radio telephone unit that interfaces to a standard telephone set acting as the transmitter and receiver between the telephone and the base station.

Operations and maintenance functions are performed at the OMC. As in PSTN-based networks, the OMC in MTSO/MSC-based networks is responsible for the day-to-day management of the radio network and provides the database and statistics for network management and planning.

The PSTN Direct Connect network is appropriate when there is capacity on the existing local or central office switch. In this case, the switch continues to provide the billing and database functions, the numbering plan, and progress tones. The MTSO/MSC architecture is appropriate for adding a fixed subscriber capability to an already existing cellular mobile network or for offering both fixed and mobile services over the same network.

Proprietary networks

While MTSO/MSC-based and PSTN Direct Connect networks are implemented using existing cellular technologies, proprietary WLL solutions are designed specifically as replacements for wireline-based local loops. One of these proprietary solutions is Nortel's Proximity I, which is used in the UK to provide wireline-equivalent services in the 3.5-GHz band. The TDMA-based system was designed in conjunction with U.K. public operator Ionica, which is the source of the "I" designation. The I Series provides telecommunications service from any host network switch, providing toll quality voice, data, and fax services. The system is switch independent and is transparent to DTMF tones and switch features.

The Proximity I system architecture consists of the following main elements:

§  Residential service system (RSS), which is installed at the customer premises and provides a wireless link to the base station

§  Base station, which provides the connection between the customer's RSS and the PSTN

§  Operations, Administration, and Maintenance system, which provides such functions as radio link performance management and billing

Residential Service System (RSS). The RSS offers two lines, which can be assigned for both residential and home office use, or for two customers in the same 2-km area. Once an RSS is installed, the performance of the wireless link is virtually indistinguishable from a traditional wired link. The wireless link is able to handle high-speed fax and data via standard modems, as well as voice. The system supports subscriber features such as call transfer, intercom, conference call, and call pick-up.

The RSS has several components: a transceiver unit, residential junction unit (RSU), network interface unit, and power supply. The transceiver unit consists of an integral 30 cm octagonal array antenna with a radio transceiver encased within a weatherproof enclosure. The enclosure is mounted on the customer premises and points toward the local base station.

The RJU goes inside the house where it interfaces with existing wiring and telephone equipment. The Proximity I system supports two 32-Kbps links for every house, enabling subscribers to have a voice conversation and data connection for fax or Internet access at the same time. At this writing, work is under way to develop systems that can handle ISDN speeds of 64 Kbps and beyond. Further developments will result in RSSs that can handle more lines per unit for medium-sized businesses or apartment blocks.

The network interface unit, mounted internally or externally, is a cable junction box that accepts connections from customer premises wiring. The unit also provides access for service provider diagnostics and contains lightening protection circuitry.

The power unit is usually mounted internally and connects to the local power supply (110/220V AC). The power unit provides the DC supply to the transceiver unit. A rechargeable battery takes over in the event of a power failure and is capable of providing 12 hours of standby and 30 minutes of talk time.

Base station. The base station contains the radio frequency equipment for the microwave link between the customer's RSS and the PSTN, along with subsystems for call-signal processing, frequency reference, and network management. This connection is via radio to the RSS and by microwave radio, optical fiber, or wireline to the local exchange. The base station is modular and can be configured to meet a range of subscriber densities and traffic requirements. The base station has several components: transceiver microwave unit, cabinet, power supply, and network management module.

The base station's dual antenna transceiver microwave unit provides frequency conversion and amplification functions. Each unit provides three RF channels, the frequency of which can be set remotely. The unit can be configured for a maximum of 18 RF channels. The antennas are available in omnidirectional or sectored configurations, depending on population densities and geographical coverage. An omnidirectional system can support 600 or more customers, while a trisectored antenna can serve more than 2000 customers. Base stations in rural areas can be sited up to 20 km from a subscriber's premises.

The base station can be configured with either an internal or external cabinet. The internal cabinet is for location in an equipment room, while the external cabinet is weather-sealed and vandal-proofed for outside locations. Both types of cabinets house the integrated transceiver system, transmission equipment, optional power system, and batteries. A separate power cabinet provides DC power to the base station from the local 110/220-V AC source. This cabinet may include battery backup with battery management capability and power distribution panel that provides power for technicians' test equipment. The network management module is the base station polls individual RSS units to flag potential service degradation. Reports include link bit error rate (BER), signal-to-noise ratio, power supply failure, and the status of the customer standby battery.

The connection from the base stations to the local exchange on the PSTN is via the V5.2 open standard interface. In addition to facilitating interconnections between multivendor systems, this interface enables operators to take full advantage of Proximity I's ability to maximize spectrum utilization through allocation of finite spectrum on a dynamic per-call basis, rather than on a per-customer basis. Concentration allows the same finite spectrum to be shared across a much larger number of customers, producing large savings in infrastructure, installation, and operations costs for the network operator.

Operations, administration, and maintenance. OA&M functions are implemented through an element manager accessed through a field engineering terminal. In Nortel's Proximity I, the element manager is built around Hewlett Packard's OpenView. Communications with the network of base stations and customer equipment is done through the Airside Management Protocol, which is based on the OSI Common Management Information Protocol (CMIP). The field engineering terminal can operate in a remote operations center, but is primarily intended for use by on-site maintenance engineers who are responsible for the proper operation of the base stations.

All the applications software in the customer premises equipment is downloadable from the element manager. This software provides the algorithms which convert analog voice signals into 32-Kbps digital ADPCM, which provides toll quality voice transmission. Other applications software includes algorithms for controlling the draw of battery-delivered power, in the event of a 110/220-V AC power failure.

Via the Air Interface Protocol, the customer equipment is able to provide the element manager with information about its current status and performance, the most useful of which are measurements taken during the transmission of speech. This allows the management system to flag performance degradation for corrective action.