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.
Asterisk
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.
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Citation Details
Title: The 2000 Elections - A Pain in the Asterisk!
Publication:Environmental Insider News (Magazine/Journal)
Date: November 28, 2000
Publisher: Environmental Insider News
Distributed by Thomson Gale
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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