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SIP Trunking with CUCM
Transcript
00:07
[Music]
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Welcome back.
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In the previous lecture we worked on advanced SIP registration with multiple softphones.
00:14
In this lecture we’ll integrate multiple Q-SYS softphones with
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a real SIP proxy using what’s called SIP trunking.
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This is useful for situations where you need to configure more than 8 softphones.
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It can also be used to talk to a proxy that does not use registration.
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In this lecture we’ll show how this is done using CUCM.
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A quick review of the basic topology of a typical VoIP deployment is shown above.
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Everything here stays the same when we are working
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with multiple softphones using trunking instead of only one.
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Here we see a network switch in the middle, a core,
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a CUCM server, and a laptop which can configure the core or run another softphone.
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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 CUCM implementation.
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The steps will be as follows:
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Biggest difference here is that there is no registration taking place.
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Without registration things happen a bit differently with the proxy.
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In this case both sides use what is called an “OPTIONS Ping” to check the status of each other.
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This is similar to a more standard network ICMP ping but using SIP signaling instead.
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Both sides will respond with an
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‘OK’ message to let the other know that they are ready to make calls.
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As mentioned before, SIP trunking is useful when you need lots of softphones in your design.
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For example, CUCM only supports 8 softphones with an advanced setup and with
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20 or 30 softphones typing in credentials would be time consuming as well!
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Here we have the CUCM side.
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The setup here is going to look very simple but there are some issues that can come up.
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In CUCM you would go to Device -> Trunk to create a new trunk.
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First, the basics. This will be a trunk and it needs a name.
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A couple things we want to note here.
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The ‘Destination Address’ will be the LAN
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interface of the core you are going to use for the softphone.
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The destination port will normally be 5060 and is the same one configured on the core.
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We use a basic SIP Trunk Profile here and a Standard SIP Profile.
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Last, and important, we have the DTMF signaling method.
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We normally use RFC2833 on the core and this should match what the core is using.
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Otherwise we won’t be able to send or receive DTMF.
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On the core side many things remain the same but there are important differences.
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Here the Username only functions as the phone number.
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The proxy IP address is where SIP signaling messages will be sent.
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‘Register with Proxy’ is not checked since with a trunk we are not doing registration.
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We can still have a backup proxy if one is needed.
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Since we are not doing registration how do both sides know that they can place calls?
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They use what is called ‘OPTIONS’ pings.
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These are not the same as ICMP messages that you might be used to and are SIP signaling messages.
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Both sides will periodically send out ‘OPTIONS’ messages
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and the other side will reply with a ‘200 OK’.
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This lets both sides know they are ready for calls.
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On the previous slide we saw an example of the OPTIONS message
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and here we see the ‘200 OK’ indicating a good status.
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If the ‘200 OK’ does not come back check connectivity between the core and the CUCM server.
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Here we see that the trunk setup failed because
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we did not see a response to the OPTIONS message that the core sent.
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Again here we need to check connectivity between the core
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and the CUCM server and make sure all IP addresses and ports are correct.
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Once we get our configuration right then at this point you should be able to
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open the softphone and see it in ‘OK’ status again.
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You should be able to place and receive a call.
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Also note again that a trunk supports more than 8 softphones
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so you could have many softphones in your design.
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You can also refer back to the basic CUCM training for reference.
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Again, nothing changes in the call itself with a trunk. It’s just how we perform the setup.
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That’s it.
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In these workshops we’ve covered the basic types of SIP setup with the core.