Telephony
 (→Session Initiation Protocol (SIP))  | 
			 (→Session Initiation Protocol (SIP))  | 
			||
| Line 76: | Line 76: | ||
|-  | |-  | ||
|rowspan="5"|[[SIP in Yate]] - this page describes the SIP protocol and some SIP features in Yate  | |rowspan="5"|[[SIP in Yate]] - this page describes the SIP protocol and some SIP features in Yate  | ||
| + | |rowspan="5"|[[SIP Client]] - how to build a SIP Client in Yate used as a Client  | ||
|-  | |-  | ||
|[[SIP Listeners in Yate]] - building a listener (TCP, UDP, TLS) "tells" Yate on which network interfaces to listen.     | |[[SIP Listeners in Yate]] - building a listener (TCP, UDP, TLS) "tells" Yate on which network interfaces to listen.     | ||
| + | |[[SIP Features Module]] - describes SIP features that are implemented in Sip Features Module and not in ysipchan  | ||
|-  | |-  | ||
|[[SIP Messages]] - allows Yate to process SIP methods by handling messages by their names.    | |[[SIP Messages]] - allows Yate to process SIP methods by handling messages by their names.    | ||
| + | |[[SIP Features Module]] - describes SIP features that are implemented in Sip Features Module and not in ysipchan  | ||
|-  | |-  | ||
|[[SIP Routing in Yate]] - configure a SIP channel or a SIP line.    | |[[SIP Routing in Yate]] - configure a SIP channel or a SIP line.    | ||
| + | |[[SIP Send DTMFs]] - this page describes DTMFs send related configuration of SIP channel.  | ||
|-  | |-  | ||
|[[SIP Configuration File]] - configuration file for sip module in Yate  | |[[SIP Configuration File]] - configuration file for sip module in Yate  | ||
| − | |||
| − | |||
| − | |||
| − | |||
| − | |||
| − | |||
| − | |||
|[[SIP Attended Call Transfer In Cluster]] - describes how to configure Yate to handle a SIP attended call transfer for an unknown call leg.  | |[[SIP Attended Call Transfer In Cluster]] - describes how to configure Yate to handle a SIP attended call transfer for an unknown call leg.  | ||
|-  | |-  | ||
| − | |[[SIP Flood Protection]] - shows how Yate provides a protection mechanism against several types of SIP flood attacks.  | + | | ||[[SIP Flood Protection]] - shows how Yate provides a protection mechanism against several types of SIP flood attacks.  | 
|-  | |-  | ||
| − | |[[SIP SBC]] - describes how Yate can be used as a SIP session border controller providing security for the internal network of the carrier.  | + | | ||[[SIP SBC]] - describes how Yate can be used as a SIP session border controller providing security for the internal network of the carrier.  | 
|-  | |-  | ||
| − | |[[SIP Generic Message]] -  the message and his syntax generated by the ysipchan module on receiving the specified requests.  | + | | ||[[SIP Generic Message]] -  the message and his syntax generated by the ysipchan module on receiving the specified requests.  | 
|-  | |-  | ||
| − | |[[xsip.generate]] - another SIP message sent by the ysipchan module requesting the transmission of a SIP request.  | + | | ||[[xsip.generate]] - another SIP message sent by the ysipchan module requesting the transmission of a SIP request.  | 
|-  | |-  | ||
| − | |[[SIP query for CNAM and LNP]] - another SIP feature used in a routing module that allows Yate to query [http://en.wikipedia.org/wiki/Caller_id|Caller Name] and [http://en.wikipedia.org/wiki/Local_number_portability Local Number Portability] databases over the SIP protocol.  | + | | ||[[SIP query for CNAM and LNP]] - another SIP feature used in a routing module that allows Yate to query [http://en.wikipedia.org/wiki/Caller_id|Caller Name] and [http://en.wikipedia.org/wiki/Local_number_portability Local Number Portability] databases over the SIP protocol.  | 
|-  | |-  | ||
| − | |[[SIP NAT]] - describes how Yate as a SIP Server can pass the voice (RTP) as a NAT.  | + | |||[[SIP NAT]] - describes how Yate as a SIP Server can pass the voice (RTP) as a NAT.  | 
|}  | |}  | ||
Revision as of 18:11, 6 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.
| About SIP | SIP Features | 
|---|---|
| SIP in Yate - this page describes the SIP protocol and some SIP features in Yate | SIP Client - how to build a SIP Client in Yate used as a Client | 
| SIP Listeners in Yate - building a listener (TCP, UDP, TLS) "tells" Yate on which network interfaces to listen. | SIP Features Module - describes SIP features that are implemented in Sip Features Module and not in ysipchan | 
| SIP Messages - allows Yate to process SIP methods by handling messages by their names. | SIP Features Module - describes SIP features that are implemented in Sip Features Module and not in ysipchan | 
| SIP Routing in Yate - configure a SIP channel or a SIP line. | SIP Send DTMFs - this page describes DTMFs send related configuration of SIP channel. | 
| SIP Configuration File - configuration file for sip module in Yate | SIP Attended Call Transfer In Cluster - describes how to configure Yate to handle a SIP attended call transfer for an unknown call leg. | 
| SIP Flood Protection - shows how Yate provides a protection mechanism against several types of SIP flood attacks. | |
| SIP SBC - describes how Yate can be used as a SIP session border controller providing security for the internal network of the carrier. | |
| SIP Generic Message - the message and his syntax generated by the ysipchan module on receiving the specified requests. | |
| xsip.generate - another SIP message sent by the ysipchan module requesting the transmission of a SIP request. | |
| SIP query for CNAM and LNP - another SIP feature used in a routing module that allows Yate to query Name and Local Number Portability databases over the SIP protocol. | |
| SIP NAT - describes how Yate as a SIP Server can pass the voice (RTP) as a NAT. | 
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
-  About MGCP
- MGCP call agent module - this module implements the Call Agent part of MGCP allowing Yate to control analog or digital Media Gateways
 
 - MGCP Features
 
-  About MGCP
 
IAX
Iax module implements IAX protocol that will send / receive trunked audio frames on UDP.
-  About IAX
- IAX - explanation how the connection works with trunked packets
 
 -  IAX Features
- Yate as IAX server and YateClient as IAX client - setup to configure Yate as a IAX Server and YateClient as a IAX client
 - Connecting To IAXTEL - setup for connecting to IAXTEL
 
 
-  About IAX
 
JINGLE
This module supports the Jingle protocol by implementing XMPP as an external component of another server.
- About JINGLE
 -  JINGLE Features
- Jabber Client With Jingle Yate Server
 - Connecting to GMail - learn how to route calls to contacts using Gmail account in accfile.conf
 
 
JABBER or XMPP (Extensible Messaging and Presence Protocol)
- About Jabber
 - Jabber Features