VoIP SIP Communication
Session Initiation Protocol (SIP) is a signalling protocol used for establishing communication sessions within an IP based network (VoIP). A session could be just as simple as a two-way phone call or it could be a more advanced multi-media based conference session. The ability to establish these sessions allows additional innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with contact lists, and IP type centrex solutions.
In recent years Voice over IP innovators (VoIP) have adopted SIP as its preferred protocol of choice for data signalling. SIP communication complies to RFC standards (RFC 3261), established by the Internet Engineering Task Force (IETF), the body responsible for administering and developing the mechanisms that operate over the Internet. SIP is still an evolving technology which is being extended further as it starts to mature.
SIP protocol is a request-response based protocol that closely resembles two other better know internet protocols, HTTP and SMTP (the protocols that power the world wide web and email); consequently, SIP sits comfortably alongside Internet based applications. Using the protocol, SIP VoIP services become another web type application and integrates easily into other internet services. SIP protocol can be viewed as a simple toolkit that service providers can use to build converged voice and multimedia based services.
The IETF work to straightforward guidelines: specify only what you need to specify. SIP protocol is very much of this mold; having been developed purely as a way to establish sessions, it does not know about the details of a session itself, it just initiates, terminates and modifies sessions. This simplicity means that SIP communication scales, it is extensible, and it sits very comfortably in different architectures and deployment scenarios.
When providing SIP phone services over VoIP there is a need for a number of different standards and protocols to be joined together - specifically to ensure transport (RTP), to authenticate users (RADIUS, DIAMETER), to provide directories (typically known as LDAP), to be able to guarantee voice quality (QoS, RSVP, YESSIR) and to interconect with today's "traditional" based telephone network (PSTN).
SIP Session Initiation Protocol (SIP protocol background)
VoIP Explained
Put simply VoIP (Voice over IP) or IP telephony is the transmission of a telephone call over IP (computer) based networks instead of traditional telephone networks or PSTN.
The Internet Protocol (IP) was originally designed for data networking, however due to the success of IP becoming a world standard for data networking it has led to it being adapted to carry telephone calls (Voice over IP or VoIP).
VoIP Definition:
Voice over IP, or VoIP, is the technology used to transmit voice conversations over an internal or external data network using IP packets (digital form); without loss in functionality, reliability or quality; and in compliance with the International Telecommunications Union specifications. The term is also used to refer to the hardware and software used to carry such calls over the network.
How is a Voice over IP call made?
A voice signal from a Voice over IP phone (or an older phone connected through a suitable VoIP adapter) is passed through a VoIP device that converts the regular telephone voice signal to a digital one so it can use a broadband internet connection where it travels to the destination equipment. The digital signal is then converted back to the original voice call.
In other words, when the originator calls a number the Voice over IP (VoIP) phone logs on to the routing server - which looks up the destination IP number that's associated with the dialled phone number - and it makes the connection. If the destination number isn't using VoIP, and doesn't have the phone number tied in with an IP number, then it is recognised that the destination number is a Public Switched Telephone Network (PSTN) phone and the call is routed through the PSTN.
SIP Protocol
VoIP SIP
VoIP SIP News
Voiceflex, the SIP solutions provider increases sales by 30% in Q1 2009.
Voiceflex Q1 2009 results show an impressive 30% increase on Q4 of 2008. The results are better than expected given the present financial climate. A mid-range single digit growth was forecast, but the orders just kept coming in.
Many factors have attributed to the growth, new manufacture approvals, a wider more diverse reseller base, the applications Voiceflex SIP trunks offer (not all SIP carriers offer the same feature set) and above all reseller confidence in the service.
The down side was a reduction in sales of their VxDSL product range, which is a fully managed voice ADSL. Voiceflex said they had expected this as ISP’s launch their own variants of ADSL2 aimed at the voice channel. The channel has a wider choice of providers with a good selection of ADSL2 routers which resellers can manage themselves.
Voiceflex are extremely pleased with the results and Q2 also looks to maintain double-digit growth. They are looking forward to the rest of the year with renewed vigour.
About Voiceflex
Voiceflex is the de facto provide advanced SIP telephony services for UK businesses. Bringing the reliability of the Internet to the telephone, Voiceflex is an advocate of SIP (Session Initiation Protocol) - the latest technology allowing voice calls to be made over the Internet. Voiceflex uses its own SIP technology offering developed completely in-house, to provide low cost, ISDN replacement lines that provide the best possible call quality, inexpensively with the flexibility that comes from using the Internet. Voiceflex can also port any telephone number regardless of geographic location. Products include SIP Trunks.
For further information contact:
Paul Taylor
Tel: 0207 440 1811
Email: ptaylor [at] voiceflex.com
