SIP Router

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Yate is a B2BUA (back-to-back user agent) that receives calls on SIP and sends them through SIP.
 
Yate is a B2BUA (back-to-back user agent) that receives calls on SIP and sends them through SIP.
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[[File:sip-router-b2bua.png]]
  
 
== Protocols used in session communication in Yate - SIP Router ==
 
== Protocols used in session communication in Yate - SIP Router ==
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===== Configure a SIP line=====
 
===== Configure a SIP line=====
  
From [[Accfile|accfile.conf]] you can configure a SIP line. In this case Yate is registers to a remote server.<br>
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From [[Accfile|accfile.conf]] you can configure a SIP line. In this case Yate registers to a remote server.<br>
  
 
This is an example:
 
This is an example:
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Configure a SIP line called [test_sip].
 
Configure a SIP line called [test_sip].
  

Latest revision as of 16:00, 13 May 2013

Yate can be used as a SIP Router that is a SIP Softswitch with his features like establishing, modifying and finalizing a session, registering and redirect just to mention a few. The implementation of SIP it is done using a VoIP SIP driver based on YASS library.

Contents

[edit] Yate - SIP router

Yate is a B2BUA (back-to-back user agent) that receives calls on SIP and sends them through SIP.


Sip-router-b2bua.png

[edit] Protocols used in session communication in Yate - SIP Router

There are various protocols used to carry data over a real time multimedia session in numerous formats like voice, video or text messages.

SIP doesn't offer any other services besides the session configuration related ones, that is why is being used along with protocols specialized on other tasks like the:

  • Real-time Transport Protocol (RTP) for data transporting and Quality of Service;
  • Session Description Protocol (SDP) for describing multimedia sessions and others.

This way SIP can provide complete functionality to the users but despite all this collaboration with other protocols it does not depend on them.

[edit] Yate features if used as SIP router

  • Besides establishing, modifying and finalizing a session, SIP can also be used to invite participants to an already existing one.
  • SIP secure uses TLS protocol for securing the signalling. This will stop man in the middle attacks and hide the called identity. Data security protocols such as SRTP rely upon a separate key management system to securely establish encryption and/or authentication keys. TLS will protect the SDP message.
  • SUBSCRIBE - Allow SIP clients to subscribe to specific events
  • NOTIFY - Notify SIP clients when an event's state changed
  • Registration allows a SIP endpoint to register with a SIP registrar so other SIP endpoints can find it. Logging into an instant message client and then showing up that is online is an example of registration.
  • If forward_sdp is set in Yate enables the entire SDP to be passed unchanged.

[edit] How to configure SIP Router in Yate

This are the configuration files for SIP:

  • ysipchan.conf the main file for configuring SIP methods
  • sipfeatures.conf configures some SIP features
  • sip_cnam_lnp.conf controls how CNAM and LNP queries are performed

For configuring users, gateways no matter the protocol used, other files are used as regfile.conf or regexroute.conf.

Using the default configuration from ysipchan.conf file, Yate will behave as a SIP server, that will listen on all interfaces or you can configure your own listeners. In Yate you can build UDP, TCP, TLS listeners.

[edit] SIP Routing in Yate

[edit] Route to a SIP channel

From a routing module (e.g regexroute.conf) you can call an sip channel like this:

;extension match rule=sip/sip:user@ip:port
^100$=sip/sip:user@ip:port

[edit] Route to a SIP line

[edit] Configure a SIP line

From accfile.conf you can configure a SIP line. In this case Yate registers to a remote server.

This is an example:

Configure a SIP line called [test_sip].

accfile.conf

[test_sip]
enabled=yes
protocol=sip
username=me
description=Test SIP account
interval=600
authname=metoo
password=1234
domain=somewhere.org
registrar=10.0.0.1:5060
outbound=10.0.0.1:5061
localaddress=192.168.0.1:5062
[edit] Route to the configured line

Route a call to the SIP line configured above:

regexroute.conf

^100$=line/100;line=test_sip


See also

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