SIP Registration with Avaya
Transcript
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[Music]
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Hi everyone, today our topic with be SIP telephony and some basic scenarios.
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In the Quantum ‘Introduction to SIP Telephony’ topic,
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we discussed the underlying technology of SIP telephony,
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but in this section we’ll look at SIP telephony specifically integrating to an Avaya system.
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We’ll go through the setup of a typical connection to an Avaya system
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and how it works with the Q-Sys core.
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In this particular case,
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we’ll explore the methods for integrating the Q-SYS softphone to an Avaya system.
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An Avaya system typically consists of two components – a session manager or SM,
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and a communication manager, or SM.
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SIP Endpoints register to the session manager and the communication manager handles
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other functions including communication to the larger PSTN network.
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We’ll test this integration by making test calls to and from Q-SYS
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from a softphone application running on a PC.
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Keep in mind the topology here is greatly simplified from an
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actual enterprise VoIP deployment,
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but the basic concepts shown here should apply to any running Avaya implementation.
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In your network you may have additional devices like routers and firewalls,
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and switches not pictured here.
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The steps will be as follows:
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In this case, devices attempt to register to the session manager.
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It checks the credentials of each endpoint and responds accordingly.
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The endpoints can make calls when and only when they’re registered with the session manager.
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There are two different methods required to interface Q-SYS to Avaya
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depending on the number of Q-SYS
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softphones required in your project and your preferred setup.
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In this workshop we are going to focus on softphone registration.
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The scenarios described here are specific to Avaya.
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Other systems have different ways of handling
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these implementations and will be discussed in other trainings.
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The steps to configure an account in Avaya for the Q-SYS Softphone are as follows:
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First we create a User profile that will be used for the Softphone.
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This will contain the authentication credentials.
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Next we create a Communication Address that includes the extension and an optional domain.
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Third we configure an Application Sequence that includes the Session Manager(s)
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being used and specifying that sip users will be using it.
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Last we configure an Endpoint Profile which assigns the extension to an endpoint
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and specifies items like the phone type.
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Let’s take a look at each of these in detail.
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Navigate to Home→Users→User Management→Manage
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Users and we see a few fields we need to configure.
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This configuration is automatically synchronized with Communication Manger.
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Let’s see what needs to be in those fields for a new user profile.
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Fill in the fields on the previous screen as shown above to configure a New User Profile.
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These credentials will be the ones we eventually use
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to configure the softphone in core manager later.
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Click the Communication Profile tab, then click
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New to define a Communication Address for the new SIP user.
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After you click ‘New’ you will see this screen. Select ‘Avaya SIP’ as the type.
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This is the default as well. Then put in the extension number and sip domain. Then click ‘OK’.
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Next, in the Session Manager Profile section, specify the Session Manager(s) and assign the
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Application Sequence to both the Originating Sequence and Termination Sequence fields.
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Think back to our SIP messages. In the Register message we had a ‘Contact’ header.
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In other SIP messages we recall that they contain SDP which has the parameters to set up our audio.
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Our last step is to configure an Endpoint Profile.
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System: Select the managed element corresponding to Communication Manager
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Profile Type: Select Endpoint
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Use Existing Stations:
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If field is not selected, the station will automatically be added in Communication Manager
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Extension: Enter extension number of the SIP user
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Template: Select a template for the type of SIP phone
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That’s it on the Avaya side.
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Let’s move on to the core manager side and see how things correspond to what we have done here.
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Now that the account is set up in Avaya, we turn our attention to the setup in Q-SYS Core Manager.
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First and foremost we must confirm the core network configuration…
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we want to make sure the correct Q-SYS NIC is connected to the VoIP network
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and the correct IP address options are chosen.
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We spent a substantial amount of time on this in the networking sections of Quantum training,
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so we’ll trust that we know how to accomplish this.
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The next step is to confirm the shared settings for all softphones.
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In this setup we will use LAN A as the VoIP interface.
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The SIP signaling port is set to 5060, which again is the default.
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This is the SIP signaling port for the core and is used for incoming calls to the core.
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The port used for outgoing calls to talk to Avaya is set later.
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SIP logging is enabled, which will be a useful troubleshooting tool if we run into problems.
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The rest of the shared settings we covered in Quantum training before
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but a quick refresh of the typical settings is shown.
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SRTP is normally disabled. DTMF INFO is disabled as well.
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Avaya does not typically use this method. The DTMF type is also fine for this application.
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STUN is also disabled.
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This applies mostly to hosted SIP solutions when
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the core is behind a firewall which is not the case here.
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Now let’s look at the individual Softphone settings.
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The ‘proxy’ field specifies the address of the Avaya session manager server.
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In some scenarios a network can have more than one Avaya server.
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In redundant proxy applications, there’s a ‘Backup Proxy’ field to point to the secondary unit.
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By default the proxy is using port 5060 but if you
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need a different port you can append it to the proxy ip with a ‘:’.
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In this case we do want to register with the proxy. Most systems require
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this level of security, including this one.
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The transport setting selects the method of SIP communication required for Avaya.
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In this case, Avaya is set to UDP.
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We now move to the individual account settings we set in Avaya.
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The ‘Username’ field represents the extension number assigned in Avaya.
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These two fields must match or we will not register.
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The Authentication ID represents the login name in the Avaya setup that we configured earlier.
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In this case it will match the extension but it could
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be configured differently depending on how the Avaya side was configured.
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At this point we should be able to go to our softphone block in our design
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and see the softphone successfully registered.
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If more than one extension is needed go back and repeat the process for another softphone.
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That’s it for Avaya setup.
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In future sessions we’ll take a look at connecting to other types of systems
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and some further information on advanced SIP features on the core.