Fundamentals of SIP

Session Initiation Protocol, SIP, is poised to continue its reshaping of your collaboration and communication network. This Fundamentals animation was produced as part of a TechWiseTV episode 66 'SIP, Session Management and Beyond.

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The Basics Of How SIP (Session Initiation Protocol) Drives Your VoIP System Design And Function

Hold on to your seat. Finally....here's an explanation of SIP (Session Initiation Protocol) in laymans terms. It's pretty basic.... but should be just what you need to understand how this VoIP application fits into your VoIP system design plans and function. After all, not everyone in the business world can talk or understand "techie speak".



SIP - Session Initiation Protocol. It is exactly that - the main purpose of SIP is to set up and tear down media (audio/video/etc data) sessions, and also to manage endpoints and other things.



SIP devices communicate (usually) on UDP port 5060. When one device wants to start a call to another, it sends an INVITE message. Included in this is the SDP, session description protocol, which explains exactly what form the data will take (audio/video/etc, what codec, etc). When they agree and are ready to start exchanging media (data), RTP (realtime transport protocol) is used to actually exchange the data. RTP functions on any range of ports, which are assigned to each endpoint. The endpoints negotiate and choose acceptable ports on each side.



Sip also does a few other thing such as REGISTER. Register allows a SIP device with a dynamic IP to recieve calls. A common use is an ATA (vonage type box) - when you plug it in, it registers to its server, and renews the registration every XXX seconds to keep the server up to date (in case its IP changes).



SIP has a handful of other functions. For example NOTIFY can be used to pass assorted data to an endpoint (many IP phones will reboot when you NOTIFY them with the data 'check-sync'). NOTIFY is also used for MWI. There is also SUBSCRIBE, which allows an extension to subscribe to notifications of the status of a voice mail box (for MWI) or an extension/channel (for BLF....(busy lamp field, the thing which makes a guy's button light up when he is on the phone).



There are a handful of other SIP functions, for example REFER (transfer), BYE (hangup), etc.



SIP has three ways of dealing with DTMF signals sent while the call is in progress:



* Inband- Send the tones as audio in the media stream. Only works with G.711 ulaw/alaw codec, other codecs will distort the DTMF.



* RFC2833- Send the tones out of band but still attached to the audio stream via RTP.



* INFO- send the tones as SIP INFO packets along the control channel.



RFC2833 is probably the most common.



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



SIP does not play nice with NAT routers, mostly because of RTP - the SDP includes the source and destination IP addresses where media should be sent to, which are not always correct.



For example - if you have an ATA behind NAT, it will use its own IP (192.168) when creating the SDP. NAT will correctly translate the header, so the packet is addressed from the network's external IP. However the contents of the packet still have a 192.168 IP as the destination, which the server cannot send media to. This commonly results in calls which work except one or both parties cannot hear each other.



There are two ways to solve this - media gateways (sip-aware router that rewrites the SDP) or more commonly, STUN (Sip Traversal Under NAT). STUN is a protocol which allows a SIP device to, with the help of a STUN server, discover its own external IP and what kind of NAT it is behind. It can then write the SDP correctly and negotiate the RTP session so the NAT will not bother it.



SIP shares many response codes with HTTP. IE- 404=extension not found, 401=unauthorized, etc.



Lastly in case you are ever looking at a SIP transcript - SIP performs authentication (where passwords are used) using digests. Thus, a typical authenticated session looks like this:



device tries to connect (INVITE)....



server responds trying....



server responds 401 unauthorized with some auth info....



device responds OK....



device tries to connect (INVITE) this time with hashed auth data....



server responds trying....



server responds ok (and other phone starts to ring)....



Hopefully the above gives you a basic understanding and arms you with enough to be able to ask the right questions....for the right reasons...at the right time.



For free help finding a SIP solution for your business simply request a no cost quote here: Business VoIP Solution

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Explanation Of SIP (Session Initiation Protocol)

SIP... or Session Internet Protocol... isn't necessarily a new communications concept. It has been around awhile although it seems to be garnering a resurgence in Telephony applications today. But... how do you explain what SIP is when asked?



Good luck with this one, trying to explain SIP to technical people is tough enough.



Here's some very simple short statements that may help...



- SIP enables telephony over the internet network



- SIP allows us to packetize and prioritize voice traffic over digital circuits.



- SIP is a way voice is packed into a digital signal that is then enabled for transfer through the internet.



- SIP digs a channel in an IP network so voice/video can flow between two (or more) places. When you finish talking, SIP shuts the channel up.



- It's an internet protocol like HTTP for web browsing, only this one is used to make a phone-like connection between computers, pda, voip-phones or other devices that can talk over the internet.



- SIP is a protocol that allows unlike mediums to communicate. All you really need to know is that SIP is the new PRI and is more cost effective from a trunking perspective.



- SIP has nothing to do with the internet.... regardless of where, when, or how voice traffic is being transmitted. f it's being sent as 0's and 1's... SIP is what differentiates voice from all other data.



- SIP enables you to eliminate the cost of maintaining two networks (POTS + Ethernet) by putting your phone traffic on your Ethernet network.



- SIP is a business-class, integrated voice and data service with connectivity provided to your IP-PBX (a telephone switch that supports voice over IP)



Or... you may explain to a non-technical person by describing the SIP VoIP operation like this:



1. Callers and callees are identified by SIP addresses.



2. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. (The most common SIP operation is the invitation).



3. SIP or VoIP is a technology that allows you to make calls between devices, be it over the local network or over the Internet (Managed or un-managed). SIP is a standards based technology that behaves very much like your old telephone line but just uses the Internet as its medium.



4. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies.



5. Users can register their location(s) with SIP servers.



6. SIP messages can be transmitted either over TCP or UDP



7. SIP messages are text based and use the ISO 10646 character set in UTF-8 encoding.



8. Lines must be terminated with CRLF.



9. Much of the message syntax and header field are similar to HTTP.



10. Messages can be request messages or response messages.



Now you are armed with some basic background in simple (or as simple as possible) terms.... to explain what SIP is and does. Should you need additional help of a more technical nature in deciding what SIP solution would work best for your given business application.... simply request assistance via Business VoIP Solution.

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