Telephony
(→Session Initiation Protocol (SIP)) |
(→Session Initiation Protocol (SIP)) |
||
Line 82: | Line 82: | ||
;[[SIP Features Module]] | ;[[SIP Features Module]] | ||
:Describes SIP features that are implemented in Sip Features Module and not in ysipchan | :Describes SIP features that are implemented in Sip Features Module and not in ysipchan | ||
− | ;[[SIP | + | ;[[SIP Methods]] |
:Allows Yate to process SIP methods by handling messages by their names. | :Allows Yate to process SIP methods by handling messages by their names. | ||
;[[SIP Routing in Yate]] | ;[[SIP Routing in Yate]] |
Revision as of 10:35, 7 November 2012
Yate is a telephony engine who's main function is to connect together call legs to external devices or to internal server resources and for that he use a variety of VoIP protocols that can be used without the need of a special hardware as SIP, H.323, IAX2 or Jingle. In the next chapters you will find more informations about configuring each telephony protocol and the features supported by Yate.
Contents |
Session Initiation Protocol (SIP)
The SIP module in Yate allows you to make calls using SIP protocol.
|
|
H323
H.323 is a standard that specifies the components, protocols and procedures that provide multimedia communication services - real-time audio, video, and data communications - over packet-based networks.
|
|
MGCP
|
|
IAX
Iax module implements IAX protocol that will send / receive trunked audio frames on UDP.
|
|
JINGLE
This module supports the Jingle protocol by implementing XMPP as an external component of another server.
|
|
JABBER or XMPP (Extensible Messaging and Presence Protocol)
|
|