The basics, such as SIP (Session Initiation Protocol) drives your VoIP System Design and Function

Keep your place. Finally .... Here is an explanation of the SIP (Session Initiation Protocol) is generally understood. It 'pretty simple .... but should only be to understand how this VoIP application for the design of the VoIP system and function to adjust the timetable. Finally, not everyone can speak or understand economics talk "techie".

SIP - Session Initiation Protocol. It 's just that - the main purpose of SIP is to set up and removal media (audio / video / data, etc.)Meetings, and end points and other administrative things.

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SIP devices communicate (usually) 5060th on UDP port will be opened at a device with a call to another, it sends an INVITE message. These will include the SDP, Session Description Protocol, which explains exactly take the form of data (audio / video / etc, what codec, etc.). When they agree and are ready to begin exchanging media (data) , RTP (Real Time Transport Protocol) used to exchange data effectively. RTP will work on anya number of ports that are assigned to each endpoint. The endpoints negotiate and define acceptable doors on each side.

The basics, such as SIP (Session Initiation Protocol) drives your VoIP System Design and Function

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Sip and do not register as something else. The registration allows a SIP device with a dynamic IP incoming calls. A common use is an ATA (Vonage box) - if you connect it, it registers its server and renews the registration every XXX seconds to keep the server up to now (when its IP changes).

SIP is a handful of other functions. For example, notificationYou can pass sorted data to an endpoint (many IP phones will be restarted when learning about them with the data of "check-sync") be. NOTIFY is used for MWI. There are also subscribe to sign an extension of the parties concerning the status of a voicemail (for MWI) or an extension / channel (for field BLF lamp ....( busy, what makes a man the light button when he is on the phone).

There are a handful of other SIP features, see the example (transfer), BYE(Hangup), etc.

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

* Toni-band as in the audio stream in the media. It only works with G.711 ulaw / alaw, codecs distort the DTMF.

* RFC2833-send the notes of the band, but still attached to the audio stream over RTP.

* The text of the INFO-known as SIP INFO packets to the control channel.

RFC2833 is probably the most common.

There 's also a number of extensionsSIMPLE (SIP Instant Messaging, Presence and extensions of the place). In short, this is a way to use SIP for Instant Messaging with the type.

SIP is not nice to play with the NAT router, mainly because of the RTP - contains the SDP, the source and destination IP addresses, where media to be sent is not always correct.

For example - if you have an ATA behind NAT, using its own IP (192.168) in preparation for the SDP. NAT is correctly translate the header, the packetDirect from the external IP network. But the contents of the package or a 192.168 IP as a goal that you can send the media server. This often results in calls not hear from one job or both parties against each other.

There are two ways to solve this - Media Gateway (SIP-aware router that writes SDP) or more often as STUN (NAT Traversal in SIP). STUN is a protocol that allows a SIP device to discover with the help of a STUN server, rightExternal IP address and what type of NAT it is behind. E 'then the SDP and write properly negotiate the RTP session, so that the NAT does not worry him.

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

Finally, if you ever watch a SIP - SIP authenticates (in which passwords are used) with digested. Thus, a typical session authenticated as follows:

Device tries to connect (INVITE )....

Server respondstry ....

Auth server responds with 401 Unauthorized some info ....

Unit will respond OK ....

Device tries to connect (INVITE), this time with hash data auth ....

Server looking for answers ....

Server responds ok (and other phone starts ringing )....

We hope the above gives you a basic understanding and your arms will be sufficient to ask the right questions .... for the right reasons ... the right time.

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The basics, such as SIP (Session Initiation Protocol) VoIP pushes your System Design and Function

Keep your place. Finally .... Here is a statement of the SIP (Session Initiation Protocol) is generally understood. It 'pretty simple .... but just to understand how this VoIP application will need to be in a VoIP system design and management plans. Since not everyone can speak to the business or understand "techie talk".

SIP - Session Initiation Protocol. It 'just the set - the main purpose of the SIP and media tearing (audio / video / data, etc.)Meetings, and to manage endpoints and other things.

SIP devices to communicate (usually) on UDP port 5060 If a device wants to establish a call to another, sends an INVITE. Included in this is the SDP, Session Description Protocol, which has exactly the form which explains the data (audio / video / etc, what codecs, etc..) If they agree and are ready to begin exchanging media (data transmission), RTP (Realtime Transport Protocol) is used for the exchange of facts. RTP will work on anya number of ports that are assigned to each endpoint. Endpoints to negotiate and select acceptable connections on each side.

SIP is something like few other records. Register, which allows a device with a dynamic IP SIP incoming calls. A common use is an ATA (Vonage box) - if you plug it in, it registers to its server and renews the registration of every XXX seconds, the server is always updated (if its IP changes).

SIP is a handful of other functions. For example, notifysorted data can be used to end at a point (with many IP phones will reboot When share it with data from 'check-sync ") are. Mail is used for MWI. There are also subscribe to an extension of the status of messages a voice mail (to subscribe to MWI) or allows an extension / channel (for the BLF lamp ....( busy, what makes a man the light button while the phone).

There are a handful of other SIP functions, refer to the example (transfer), BYE(Hangup), etc.

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

* Send the inband tones as an audio stream in the media. It only works with G.711 ulaw / alaw codec, other codec distort the DTMF.

* RFC2833-send the notes of the band, but still attached to the audio stream over RTP.

* INFO to send signals, such as SIP INFO packets for the control channel.

RFC2833 is probably the most common.

There are also a number of extensions, calledSIMPLE (SIP Instant Messaging, presence and extensions of the place). In short, this is a way to use SIP for Instant Messaging type used.

SIP does not play Nice with NAT router, contains mainly because RTP - the SDP, the source and destination IP addresses, where media should be sent which are not always correct.

For example - if you have an ATA behind NAT will use your IP (192.168) during the creation of the SDP. NAT correctly translate the header, the packet isdirected by external IP network. But the contents of the package or a 192.168 IP as a goal that the server can not send to the media. This often calls that can save one or both parties do not listen to work together.

There are two ways to solve this - Media Gateway (SIP-aware router that SDP) writes, or more frequently STUN (NAT Traversal Under SIP). STUN is a protocol that a SIP device with the help of a STUN server to discover their own permitsExternal IP and what is behind NAT. You can then correct the SDP and the negotiation of the RTP session, so that NAT will not interfere.

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

Finally, if you ever look at a SIP - SIP authenticates (if passwords are used) with digested. This is typical of an authenticated session as follows:

Device tries to connect (INVITE )....

Server respondstry ....

Auth server responds with 401 Unauthorized some info ....

Unit will respond .... OK

Device tries to connect (INVITE), this time with data authentication hash ....

Server responds trying ....

Server responds ok (and the phone starts to ring )....

I hope the above gives you a basic understanding and arms enough to put in a position to ask the right questions .... for the right reasons ... at the right time.