General Questions and Answers
This FAQ section is for general and commercial questions about SIP. There are technical and troubleshooting FAQ's in the Test Area.
General Questions
General Answers
What is the SIP Center?
The SIP Center is a portal for the commercial development of SIP software. Serving both the SIP community and the wider industry, the SIP Center offers comprehensive technical and market resources as well as an environment for the testing of SIP implementations.
What is the SIP Forum?
The SIP Forum is also active in promoting SIP. Its principal purpose is to share information amongst active parties within a working group model. www.sipforum.org
I want to get involved in a SIP trial/pilot. What should I do next?
Send an email to The Editor to let us know what you are looking to achieve and we'll see if we can help you set up a trial/pilot.
I want to buy a SIP product. What should I do next?
Send an email to The Editor and we'll see if we can help you identify a suitable supplier. Alternatively take a look at the SIP product list.
Which companies are involved in the standard?
Take a look at the SIP product list for those companies that are active in development of the standard.
Why was this protocol developed?
To address the issue of multiple standards approaches to different media types and reach a common ground within the IETF standards area - maximising on relevant standards already addressing key issues and building on standards that were already in place. In addition, there are killer applications associated with IP, email and the web for example, that use simple protocols to deliver on advanced features. SIP is being developed to allow IP Telephony to become a part of this world of simple protocol interactions and advanced applications.
There is plenty of information covering this in the About SIP section.
What does SIP offer?
The integration of media forms into one integrated service capability. SIP allows programmers to be able to develop web applications that use voice as never before. The showcase section of the SIP Center outlines some possibilities.
What can SIP do for my business?
SIP will impact on the ease with which basic business processes are fulfilled. Linking the telephony, video and web world into an integrated media to be manipulated for business efficiency. Calls become media calls where information exchange can occur in any form and in multiple forms. Aspects of desktop integration ensure that the IP Telephony and the IP world as a whole integrate with a common group of similar standards.
The showcase section of the SIP Center outlines some possibilities.
What SIP services are available?
SIP services are emerging from lab trials and into commercial deployments. We list Service Providers that are currently active. If you know of, or are deploying, any other services then let us know.
Who developed SIP?
SIP is an IETF driven initiative and development. Activity has been centred at Columbia University.
Who is using SIP today?
The SIP Center follows Service Providers active in SIP. Take a look at what the market is doing in the Service Provider section.
If you know of, or are deploying, any other services then let us know .
Who is developing SIP products today?
The Products and Services section provides a list of many of the companies. This list is not exhaustive as the number is increasing all the time. Let us know if you are developing a product.
Where does SIP fit with H323?
H.323 embraces the more traditional circuit-switched approach to signaling based on the ISDN Q.931 protocol and earlier H-series recommendations. However, SIP utilises the more lightweight Internet approach based on HTTP. SIP addresses some of the perceived shortcomings of H.323. For example, establishing an H.323v1 call can take around a dozen packets and about 6 to 7 round-trip times, depending on how connections and packets are overlapped. For a modem connection, where transmission delays are substantial, setting up an H.323 call can take several seconds. However, setting up a SIP call via UDP takes 1.5 round trip times and four packets. H.323v1 allows to address either hosts directly (but without user name), or go through an alias. All aliases have to be resolved through a gatekeeper. SIP destination and forwarding addresses can be any URL, including mailto, phone, H.323 and http URL, affording flexibility in combining SIP with other signaling protocols. Also, email-like names can be mapped by any device on the Internet.
There is extensive work being undertaken to ensure that SIP and H.323 deployments interwork.
For more information on SIP and H.323 see the relevant section in About SIP.
Why should I use SIP rather than H.323?
H.323 was originally intended to set up video conferences across LANs by the International Telecommunications Union (ITU). As H.323 was in existence before SIP there was some early adoption of this protocol for Internet Telephony. Since those days SIP has become well established. Today it is widely recognised that SIP is simpler and more extensible than H.323. This is a natural result of SIP being developed in the fast changing environment of the Internet by the Internet Engineering Task Force (IETF). SIP has also been designed to inter-operate with other IETF protocols, such as HTTP and SMTP, that provide the core of the Internet. This close relationship means that SIP is uniquely placed to provide new and exciting services that combine telephony and Internet technologies. As these new services will drive the creation of next-generation communication networks, most major vendors have now announced their support for SIP.
For more information on SIP and H.323 see the relevant section in About SIP.
How does SIP relate to MGCP/Megaco?
MGCP/Megaco and SIP are not peers; they can and will co-exist in converged networks.
Though MGCP/Megaco can control end-points, the services thus enabled are very limited. SIP, therefore, is required to control end-points and also to communicate between softswitches. MGCP/Megaco is used to internally control the media gateway.
This topic is dealt with in the relevant About SIP section.
What is a Softswitch?
A softswitch sits at the interconnect point between an IP network and other networks and controls the media gateways using a master/slave protocol such as MGCP. It also translates signalling protocols from one environment to another (e.g SIP to SS7). Softswitches are also known as Media Gateway Controllers (MGCs) and Call Agents.