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Archive for the ‘SIP’ Category

SIP vs H.323

Posted by kathayat on September 26, 2007

….H.323 is the vertically integrated suite of protocols that addresses a broad range of IP telephony issues, including such things as codec, terminal registration, call control, address translation, administion control and call authorization. In many cases there is no clear seperation of the responsibilities between these H.323 protocol elements. It is not uncommon for the service to require the interactions among a number of them.

….SIP on the other hand was designed to do nothing more than support session setup and relies on the other, unspecified, protocols and applications that take care of everything else. SIP’s modularity let it work evern with H.323

….SIP reuses the existing internet technology, for example,  URLs, MIME and DNS, that makes the SIP smaller. At the same time SIP can more easily be integrated with the existing internet applications because SIP’s syntax is closely modeled on that of HTTP (text based and thers features…)

….Softwares that works with text-basesd protocol are generally less expensive to develop and easier to debug. Also, while it’s been claimed that binary protocols take up fewer bytes than text protocols, in practive it is often not so…[Ref. Stephen M. Muller -Book: API and PROTOCOLS for convergent network services page no. 254] Furthermore, space efficiency may not be an important criterion for the protocols that exchage ony few intermittent messages (many signalling protocols).

….Complexicity and scalability issues  – SIP is better

….Future of H.323 will be as basic access technogy for the IP telephony. [Ref. Muller book]

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What is SIP

Posted by kathayat on September 25, 2007

….IETF standards track application layer protocol for establishing, modifying and tearing down sessions whose participants are connected directly or via gatewat to a network.

….Key part of the communication system is finding the call participants and contacting them. The problem is made even more interesting if you assume pasticipants may move from place to place, changing their locations and the addressable equipments they are using. Add to this notion that calls need not be restricted to a single voice stream but may involve multiple streams of voice media. Then consider that many – even thousands- of participants might be involved in that call joining and living in a constantly changing topology. Puts all these together and there is obviously a need for some sort of protocol to deal with generalised sessions. SIP fills this rols.

….SIP supports basic four functions

  1. User locations – translating the users name (email or phone) into current network address, keeping track of the users location as it moves to different locations in the network
  2. Feature negotiation – Ensure that all participants in a session agree on the features to be supported among them
  3. Call management – adding, dropping, transfering, on hold
  4. Feature modification – chaning the feature of the session while the session is in progress

…. There are some assumptions SIP follows

  1. SIP should be scalable
  2. SIP should reuse as many existing protocols and protocol design concepts as possible rather than inventing new ones
  3. SIP should maximize the interoperability

…. SIP Messages

  • Like HTTP, SIP is a text-based protocol with request and response messages
  • SIP message consists of  – request/response line followed by header lines and optional body
  • Format of the request/response line is SIP-Method/SIP-Version Req-URI/Status-Code SIP-Version/Reason-Pharse
  • SIP Methods 
    • INVITE, ACK. OPTIONS, BYE, REGISTER, CANCEL
  • SIP Request URI –
    • sip:abcd@someplace.com
    • tel:123456@voidservice.com
  • SIP Headers – keyword:value

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