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The basis of how sip (session initiation protocol) VoIP system gives your design work


Added: 13-06-2005
Author: Michael Lemm
Category: VoIP 		← Back | ↓ Related | Home →


Hold on to your seats. .... Here are the details finally SIP (Session Initiation Protocol) in laymans terms. It's pretty basic .... but you should only need to understand how this fits in VoIP applications VoIP system design and planning your work. Afterall, not everyone in the business world can speak or understand the tech talk.

SIP - Session Initiation Protocol. It is exactly that - the primary purpose of SIP is to build and destroy the media (audio / video / data, etc.) session, and also manages the endpoint and others.

SIP devices to communicate (usually) on UDP port 5.060. When someone wants to start a call to another device, it sends an invitation message. Included in this is the SDP, the session description protocol, which describes what is needed, especially data (audio / video / etc, what codecs, etc.). When they are ready to agree to start sharing the media (data), RTP (realtime transport protocol) is used to share data is correct. Performance of different RTP port, which is assigned to each endpoint. The endpoints negotiate and select a port that is acceptable to each side.

Sip also doing some other things such as REGISTER. List of SIP enables the device to receive dynamic IP calls. Normal use is ATA (Vonage box type) - when you install, it registers with the server and update the registration every XXX seconds to keep the servers up to date (in terms of changes in IP).

SIP has several other functions. Tell instance can be used to pass data to a variety of endpoints (IP many will reboot when their numbers Notify check-Sync). Tell also be used for MWI. There is also SUBSCRIBE, which allows an extension to join the communications status of the voice box (for MWI) or extension / channel (for busy ....( BLF light field, which makes a man a light switch when he was on the phone).

There are several other SIP functions, such as REFER (transfer), bye (not ngup), etc.

SIP has three ways to deal with the DTMF signal sent during a call in progress:

* Send Inband-ton as an example in the media stream. Only works with law G.711 / alaw codecs, DTMF codec would be misleading.

* RFC2833-Sending out tons of bands, but still attached through the RTP audio stream.

* INFO-send the SIP INFO tons as a package from the control channel.

RFC2833 may be normal.

There is also a set of extensions called SIMPLE (SIP Instant Messaging, Presence, Location and Extensions). Set course, this is a way to use SIP for instant messaging type used.

SIP Nat doesn't play well with routers, especially in RTP - the SDP includes the source and destination IP address where media will be used, which is not always accurate.

For example - if you have an ATA Nat back, will use their own IP (192.168) while creating the SDP. Nat header will translate accurately, so that the package is addressed from outside the IP network. But what is still 192.168 IP packet as the destination, the server can not send media. These results, which usually work only one call or the parties can not hear each other.

There are two ways to solve this problem - the media gateway (sip-aware router that rewriting SDP) or more generally, stun (Under Sip Nat Traversal). Fainting is a protocol that allows SIP devices, with the help of the stun server, went out to find its own IP and what type of Nat again. Can then write accurately and SDP session to discuss the Nat RTP so it will not bother him.

SIP shares many HTTP response code. IE-404 = extension not found, 401 = invalid, etc.

End if you've seen the transcript SIP - SIP perform authentication (where the password is used) using the digest. Thus, typical authenticated session looks like this:

device is trying to connect (CALLS )....

server response to try ....

Invalid server response 401 Auth and some info ....

response devices OK ....

device is trying to connect (CALLS) this time for the data Shed Auth ....

server response to try ....

ok server response (and others ).... phone started ringing

Hopefully this gives you the basic knowledge and hands enough to ask the right questions .... accurate because ... time.



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