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SIP based VoIP service is on path to become one of the most significant protocols since HTTP and SMTP
By Nik On June 9, 2011 · In QoS - Quality of Service, SIP protocol, Uncategorized, VoIP calls quality
Understanting SIP
The growing thirst among communications providers,their partners and subscribers for a new generation of IPbased services is now being quenched by SIP – the SessionInitiation Protocol. An idea born in a computer sciencelaboratory less than a decade ago, SIP is the first protocolto enable multi-user sessions regardless of media contentand is now a specification of the International EngineeringTask Force (IETF).Today, increasing numbers of carriers, CLECs and ITSPsare offering such SIP-based services as local and long distance telephony, presence & Instant Messaging, IPCentrex/Hosted PBX, voice messaging, push-to-talk, richmedia conferencing, and more. Independent software vendors (ISVs) are creating new tools for developers tobuild SIP-based applications as well as SIP software forcarriers’ networks. Network equipment vendors (NEVs)are developing hardware that supports SIP signaling andservices. There is a wide variety of IP phones, User Agents,network proxy servers, VOIP gateways, media servers and application servers that all utilize SIP. Gradually, SIP is evolving from the prestigious protocols itresembles — the Web’s Hyper Text Transfer Protocol(HTTP) formatting protocol and the Simple MailTransfer Protocol (SMTP) email protocol — into apowerful emerging standard. However, while SIP utilizesits own unique user agents and servers, it does not operatein a vacuum. Comparable to the converging of themultimedia services it supports, SIP works with a myriadof preexisting protocols governing authentication,location, voice quality, etc.
This is a high-level overview of what SIP isand does. It charts SIP’s migration from the laboratory to the marketplace. It describes the services SIP provides andthe initiatives underway that will spur its growth. It alsodetails the key features that distinguish SIP among protocols and diagrams how a SIP session takes place.
A New Generation of Services
Flexible, extensible and open, SIP is galvanizing the powerof the Internet and fixed and mobile IP networks to create anew generation of services. Able to complete networkedmessages from multiple PCs and phones, SIP establishessessions much like the Internet from which it was modeled.
In contrast to the longstanding International TelephonyUnion (ITU) SS7 standard used for call setup andmanagement and the ITU H.323 video protocol suite,SIP operates independent of the underlying networktransport protocol and is indifferent to media. Instead, itdefines how one or more participant’s end devices cancreate, modify and terminate a connection whether thecontent is voice, video, data or Web-based. SIP is a major upgrade over protocols such as the MediaGateway Control Protocol (MGCP), which converts PTSN audio signals to IP data packets. Because MGCP isa closed, voice-only standard, enhancing it with signalingcapabilities is complex and at times has resulted incorrupted or discarded messages that handicap providersfrom adding new services. Using SIP, however,programmers can add new bits of information to messageswithout compromising connections. For example, a SIP service provider could establish anentirely new medium consisting of voice, video and chat.With MGCP, H.323 or SS7, the provider would have towait for a new iteration of the protocol to support the newmedium. Using SIP, a company with locations on twocontinents could enable the medium, even though the gateways and devices may not recognize it.
Moreover, because SIP is analogous to HTTP in the way itconstructs messages, developers can more easily and quicklycreate applications using popular programming languagessuch as Java. Carriers who waited years to deploy callwaiting, caller ID and other services using SS7 and theAdvanced Intelligent Network (AIN) can deploy premiumcommunications services in just months with SIP. This level of extensibility is already making its mark ingrowing numbers of SIP-based services. Vonage, a serviceprovider targeting consumer and small business customers,delivers over 20,000 lines of digital local and long distance calling and voice mail to over customers using SIP. Deltathree, which provides Internet telephony products,services and infrastructure for service providers, offers a SIP based P-to-Phone solution that lets PC users call anyphone in the world. Denwa Communications, whichwholesales voice services worldwide, delivers PC to PC and Phone to PC caller ID, voice mail as well as conferencecalling, unified messaging, account management, selfprovisioning and Web-based personalized services using SIP While some pundits predict that SIP will be to IP whatSMTP and HTTP are to the Internet, others say it couldsignal the end of the AIN. To date, the 3G Community hasselected SIP as the session control mechanism for the nextgeneration cellular network. Microsoft has chosen SIP for itsreal-time communications strategy and has deployed it inMicrosoft XP, Pocket PC and MSN Messenger. Microsoftalso announced that its next version of CE.net will include aSIP-based VoIP application interface layer, and is committedto deliver SIP-based voice and video calls to consumers’ PCs.In addition, MCI is using SIP to deploy advancedtelephony services to its IP communications customers.Users will be able to inform callers of their availability andpreferred method of communication, such as email,telephone or Instant Message. Presence will also enableusers to instantly set up chat sessions and audioconferences. With SIP, the possibilities go on and on.
History
SIP emerged in the mid-1990s from the research ofHenning Schulzrinne, Associate Professor of the Department of Computer Science at ColumbiaUniversity, and his research team. A co-author of theReal-Time Transport Protocol (RTP) for transmitting realtime data via the Internet, Professor Schulzrinne also cowrote the Real Time Streaming Protocol (RTSP) — aproposed standard for controlling streaming audio-visualcontent over the Web. Schulzrinne’s intent was to define a standard for Multiparty Multimedia Session Control (MMUSIC). In 1996,he submitted a draft to the IETF that contained the key elements of SIP. In 1999, Shulzrinne removed extraneouscomponents regarding media content in a new submission,and the IETF issued the first SIP specification, RFC 2543.While some vendors expressed concerned that protocolssuch as H.323 and MGCP could jeopardize theirinvestments in SIP services, the IETF continued its workand issued SIP specification RFC 3261 in 2001.The advent of RFC 3261 signaled that the fundamentals of SIP were in place. Since then, enhancements tosecurity and authentication among other areas have beenissued in several additional RFCs. RFC 3262, forexample, governs Reliability of Provisional Responses.RFC 3263 establishes rules to locate SIP Proxy Servers.
RFC 3264 provides an offer/answer model and RFC 3265determines specific event notification. As early as 2001, vendors began to launch SIP-based services.
Today, the enthusiasm for the protocol isgrowing. Organizations such as Sun Micro systems’ Java Community Process are defining application programinterfaces (APIs) using the popular Java programminglanguage so developers can build SIP components and applications for service providers and enterprises. Mostimportantly, increasing numbers of players are entering the SIP marketplace with promising new services, and SIP is on path to become one of the most significant protocols since HTTP and SMTP.
The SIP Advantage: Open, Extensible Web-Like Communications
Like the Internet, SIP is easy to understand, extend and implement. As an IETF specification, SIP extends theopen-standards spirit of the Internet to messaging,enabling disparate computers, phones, televisions andsoftware to communicate. As noted, a SIP message is verysimilar to HTTP (RFC 2068). Much of the syntax inmessage headers and many HTTP codes are re-used.Using SIP, for example, the error code for an address notfound, “404,” is identical to the Web’s. SIP also reuses the SMTP for address schemes. A SIP address, such as “sip:guest@voip-sip.org, has the exact structure as an e-mail address. SIP even leverages Web architectures, suchas Domain Name System or Service (DNS), makingmessaging among SIP users even more extensible.Using SIP, service providers can freely choose amongstandards-based components and quickly harness newtechnologies. Users can locate and contact one another regardless of media content and numbers of participants.SIP negotiates sessions so that all participants can agree onand modify session features. It can even add, drop or transfer users. However, SIP is not a cure-all. It is neither a session description protocol, nor does it provide conferencecontrol. To describe the payload of message content andcharacteristics, SIP uses the Internet’s Session Description Protocol (SDP) to describe the characteristics of the enddevices. SIP also does not itself provide Quality of Service(QoS) and interoperates with the Resource Reservation Setup Protocol (RSVP) for voice quality. It also works with a number of other protocols, including the Lightweight Directory Access Protocol (LDAP) for location, the Remote Authentication Dial-In User Service(RADIUS) for authentication and RTP for real-timetransmissions, among many others.
SIP provides for the following basic requirements incommunications:
1. User location services
2. Session establishment
3. Session participant management
4. Limited feature establishment
An important feature of SIP is that it does not define thetype of session that is being established, only how itshould be managed. This flexibility means that SIP can beused for an enormous number of applications andservices, including interactive gaming, music and video on demand as well as voice, video and Web conferencing. There are are some of other SIP features:
TO BE CONTINUED /// …
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