Introduction
When using the FCSDK there are a number of different call flows that are supported but could require different configuration of the FCSDK and it's users to support them.
This document will attempt to define each of the call flows and discuss the configuration that would be needed to support them.
Sample Application Configuration
As most initial tests a customer run are carried out using the FCSDK Sample Application the first thing to consider is how to configure user for use with the sample application.
The users to be used to login to the Sample Application are defined in the fusionwebapp-db.xml which is supplied are part of the FCSDK installation.
For each user who will be able to make audio and video calls the following section would need to be added to the xml file
<user>
<name>user1</name>
<password>123</password>
<inboundCallingEnabled>true</inboundCallingEnabled>
<outboundDestinationPattern>all</outboundDestinationPattern>
<sipUser>1231</sipUser>
<sipDomain>192.168.8.96</sipDomain>
<authUser>1231</authUser>
<authRealm>192.168.8.96</authRealm>
<authPass>123456</authPass>
</user>
Once all the users have been added to the xml file it must then be deployed to the FAS installation. For details instructions on how to deploy this artifact please refer to the Installing FCSDK document provided with the FCSDK release.
FCSDK User to FCSDK User
To support a call between two FCSDK users each of the users must register with the FCSDK.
When using the sample application this is done automatically when you login.
FCSDK & CUCM Integration
For a full description on how to configure the FCSDK and CUCM please read the following document
This document provide details configuration information for both CUCM 8.6 and CUCM 9.6.
CUCM SIP Normalization Script
When CUCM send a request to the FCSDK it is possible that the FCSDK will return with a 404 Not Found even though the user would appear to have successfully registered with the FCSDK.
In closer inspection of the incoming Invite request you will probably notice that the SIP address in the Request Uri contains a port number of 5060
e.g. INVITE sip:1001@registrar.fcsdk.cafex.com:5060 SIP/2.0
In the SIP specification the rules for matching addresses state that a SIP address with a port number and a SIP Address without a port number do not match. Therefore the port number needs to be stripped of the end of the Request Uri before it leaves the CUCM and this can be achieved by adding the following SIP Normalization Script.
M = {}function M.outbound_INVITE(msg)
local method, ruri, ver = msg:getRequestLine()
local uri = string.gsub(ruri, "@(.*):%d+", "@%1")
msg:setRequestUri(uri)
end
return M
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