Transcript
Basic SIP Telephony
19m 56s
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Hi everyone,
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today our topic with be basic SIP telephony integration with CUCM.
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SIP integration can be one of the most challenging aspects of Q-SYS.
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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 from a more practical angle.
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In this lecture we’ll integrate the Q-SYS softphone with a real SIP
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proxy to get a deeper understanding
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of the typical setup of the proxy itself
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and what we need to know about that setup to be successful in registering and making calls.
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In this particular case,
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we’ll explore the methods for integrating the Q-SYS softphone
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to Cisco Unified Communications Manager, or CUCM.
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We’ll test this integration by making test calls to and from a
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Q-SYS system from a softphone application running on a PC.
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Keep in mind the topology here is greatly simplified
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from an 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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In this diagram we have a couple of basic devices.
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A server running Cisco Call Manager (CUCM), a core, a laptop running a softphone,
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and a network ethernet switch connecting it all together.
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In your network you may have additional devices like routers and firewalls not pictured here.
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The steps will be as follows:
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The first step for most SIP telephony endpoints is to register with the SIP proxy.
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In this case, CUCM 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 call manager.
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We’ll discover that different methods are required to interface Q-SYS to CUCM
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depending on the number of Q-SYS softphones required in your project.
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In this workshop we are going to focus on single softphone registration.
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In future workshops we’ll address multiple softphones and SIP trunking with CUCM.
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The scenarios described here are specific to CUCM.
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Other systems have different ways of handling these implementations
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and will be discussed in other trainings.
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The steps to configure an account in CUCM for the Q-SYS Softphone are as follows:
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First we create the Security profile that will be used for the Softphone.
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Next we create the End User profile that includes the
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credentials the softphone will use to register.
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Finally we create the phone definition which
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consists of a Device and an associated Directory Number.
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The Directory Number is the phone number used for the softphone. Let’s look at each step in detal.
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When one and only one softphone is required in a given Q-SYS design,
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it is typically registered as a ‘Third-party SIP Device (Basic).
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Each SIP endpoint in CUCM requires licensing credits, and this is the least expensive option.
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The very first part of defining the softphone in CUCM is to create a ‘Phone Security Profile’.
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The items to note here pertinent to the Q-SYS configuration
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are the Name, the Transport Type, Digest Authentication, and SIP Phone Port.
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We’ll go over these one by one.
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First the security profile needs a name.
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This name can be whatever you want but it will be used later so remember the name you used.
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The transport type determines whether Q-SYS should use the UDP
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or TCP network transport for SIP messages to and from CUCM.
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Remember this determines the behavior of the signaling plane, the call control messages,
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but not the audio transport used to send and receive audio.
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Digest Authentication is the name of the method the Q-SYS softphone uses
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to validate its username and password with CUCM.
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This is the only authentication method supported by Q-SYS.
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The SIP Phone Port sets the UDP/TCP port number used to interface to the Softphone.
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Port 5060 is the standard port number, but it can be manually changed by the system administrator.
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Our next step is to configure an End User.
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This screen allows us to configure a User ID and Password for authentication.
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The ‘User ID’ in the end user setup corresponds
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to the authentication username in the Q-SYS softphone setup.
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The tricky part is the password immediately below this user ID is NOT the authentication password.
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It’s also NOT the PIN below that.
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The ‘Last Name’ field doesn’t really do anything but it needs to be filled
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in so we can just use ‘Q-SYS’ as the last name.
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Instead, the authentication password is set in the ‘Digest Credentials’ field.
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That may not seem incredibly intuitive, but remember that we enabled “Digest Authentication,”
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so it makes sense that the softphone is using digest authentication to register.
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The final step is to configure a phone device of the correct type tied to that user account.
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We see here the device is set up as a basic third-party SIP device.
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There are a large number of settings on the Device screen so if following along on your own CUCM
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scroll down to see the settings above.
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Here we see the device type again along with the user ID in the ‘Owner’ field.
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The ‘Owner User ID’ is the same as you configured earlier in the End User config
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and also corresponds to the Authentication Username on the core which we’ll see later.
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There is also an entry for ‘Device Pool’ here which is set to Default.
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This is a more advanced setting but it controls which CUCM servers the device can register to
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and also access to things like media resources.
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Check with your CUCM administrator
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regarding appropriate Device Pool settings for your implementation.
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In the Protocol Specific Information at the bottom of this page,
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we assign the device security profile we created before.
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The SIP Profile controls all of the basic SIP parameters for this phone.
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Most of the time those do not need to be modified unless requested by a network administrator.
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The ‘Standard SIP Profile’ should work with our softphone but if needed you can copy the standard
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profile and create a separate ‘Q-SYS SIP Profile’
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and assign it instead. We also specify the digest user we created for this account.
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We also have a field here for ‘Media Termination Point Required’.
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This is for scenarios where you might be using G.729 to interface to CUCM
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but CUCM is using G.711 to talk to the wider telephone network.
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This is called transcoding and if calls aren’t working it’s something to check.
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This might even be required if different DTMF methods are being used in the network.
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When you are done with the configuration you should click ‘Save’ and ‘Apply Config’.
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Otherwise this configuration could be lost and is not active yet.
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The next step is to associate a directory number with a device.
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From the top of the Device screen click on ‘Add a new DN’ on the upper left.
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The directory number is the first field and represents the phone extension.
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This will normally be a 4 digit number but depends on your configuration.
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In the ‘Associated Devices’ section you would use the MAC address of
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the device from our previous step. CUCM sometimes fills this in automatically.
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Now if you go back to the device you will see the directory number associated with the device.
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Now that the account is set up in CUCM, 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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We saw this in the CUCM setup.
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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 CUCM is set later.
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SIP logging here is enabled, which will be a useful troubleshooting
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tool if we run into problems. By default this is disabled.
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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.
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DTMF INFO is disabled as well.
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CUCM does not typically use these methods.
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The DTMF type is also fine for this application. 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 CUCM server.
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In some scenarios a network can have more than one CUCM 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.
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Most systems require this level of security, including this one.
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The transport setting selects the method of SIP communication required for CUCM.
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In this case, you’ll remember that CUCM was set to use UDP.
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This could also be TCP, depending on the configuration.
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We now move to the individual account settings we set in CUCM.
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The ‘Username’ field represents the subscriber number assigned in CUCM.
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These two fields must match or we will not be able to register.
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The Authentication ID represents the digest username in the end user account setup.
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Remember, the password represents the digest credentials in the end user account setup,
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not the password or PIN fields.
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This is very important to remember.
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….And voila! Provided all those items are correct, the softphone is registered!
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Success! Wait! What? CUCM tells us in the softphone status we have a fault.
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This leads us to an important piece of trivia about CUCM.
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The Authentication username is case sensitive. Note the cases shown do not match.
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Correcting this will finally result in a successful registration!
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We covered the details of how registration occurs during Quantum training
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but as a refresher here is an example of a successful registration
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that you would see in Wireshark if you grabbed a packet capture from the core.
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Once we correct this, the ‘OK’ status indicates the softphone is registered
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and ready to make calls.
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We see green in the softphone status block and know that’s good.
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Now let’s try to make a test call.
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Now we can dial our computer softphone at extension 2001.
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The softphone is answered and we see the softphone in the connected state.
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We hang up and the call disconnects as expected. The test call was a success!
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Let’s look at the exchange between the different peers who participated in the call.
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Sometimes everything looks correct with our configuration in the core
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and CUCM and things still don’t work!
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There are a few other common things to recheck.
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If you go back to CUCM make sure that under the Device configuration
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that we didn’t miss configuring a digest user.
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It’s down pretty far in the configuration so sometimes it’s easy to miss.
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Also go back and check again that ‘Enable Digest Authentication’ is checked.
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Also easy to forget and nothing will work if the box isn’t checked.
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When we look at this interaction, it’s easy to see the role of the proxy server in call setup.
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We see an interaction between Q-SYS and CUCM.
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Then we also see an interaction between CUCM and the peer we’re calling.
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The key takeaway here is that the proxy server acts as the traffic-cop between the peers.
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The direct interaction between them comes only after the session is negotiated by CUCM.
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Let’s analyze the traffic flow from end to end to understand the process:
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At the very first we see the Q-SYS softphone
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INVITE the destination to a call by contacting CUCM.
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CUCM then issues a challenge to authenticate
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before passing along the INVITE to the destination.
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The Q-SYS softphone issues a new INVITE message with the encrypted password in response.
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This is where we’ll start charting the exchange between all the devices involved.
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The new INVITE message is received by CUCM.
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It includes the Session Description Protocol, or SDP,
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information so that CUCM knows what audio formats (codecs) it supports.
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SDP also contains information about which IP addresses will be sending and receiving RTP
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and which network ports to use for RTP.
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If SDP is included in the INVITE this is called the early offer method.
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CUCM in turn INVITEs the far end, extension 2001.
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Since 2001 is another phone registered with CUCM,
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it knows how to reach that peer where the Q-SYS Softphone would not.
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CUCM replies to Q-SYS to say TRYING… which means ’I’m working on that’
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The destination in turn replies with ‘I’m working on it’, and then with RINGING,
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to let CUCM know that it’s prompted the user of that phone to answer.
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CUCM then passes that along so that the Q-SYS softphone
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can notify the user that the far end is ringing.
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This notification is also what triggers the caller to hear ringing.
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The destination responds with an OK…this includes the SDP
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information about which codec the call will be using for that leg of the call.
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The leg between CUCM and the destination
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may not use the same codec that is used between the Q-SYS Softphone and CUCM.
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This is called transcoding.
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CUCM compares the codec list to our initial offer in the INVITE.
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It sends a message with the highest-priority match for both peers.
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Q-SYS then Acknowledges that codec selection and SDP information.
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Amongst other information included in the Session Description Protocol is the contact information
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(IP addresses, port numbers, etc for the other end of the call.
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This now allows the peers to exchange audio directly with each other.
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The peers start to exchange audio and begin the call in earnest.
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This isn’t included in the flow diagram, as the capture was taken on the machine hosting CUCM.
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These packets would only be seen on a capture from Q-SYS or the destination.
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Note that any in-call DTMF would be handled as RTP EVENT type messages directly between peers.
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They would show up in the RTP flow as well.
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After a few seconds in call, the Q-SYS softphone ends the call. We see this as a ‘BYE’ message.
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Note this goes back to CUCM rather than directly to the far end of the call.
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CUCM then forwards the BYE message to the far end. If the other side hangs up,
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the BYE messages would flow in the other direction.
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The far end responds with an OK that is also passed from CUCM to the Q-SYS softphone.
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Now we’ve seen the SIP account created in CUCM and Q-SYS,
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and we’ve seen how the data from the CUCM account setup fits in the Softphone configuration.
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We’ve conducted a simple analysis of the registration process
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and the process of making a call.
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Next let’s take a look at some other issues that can occur.
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Other status messages on your softphone may appear.
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Two common ones are ‘Registration Timeout’ and ‘Service Unavailable’.
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‘Registration Timeout’ means that the core did not get a response from CUCM.
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This might be a network issue or the wrong IP address.
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‘Service Unavailable’ means that registration couldn’t happen.
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This might mean that digest authentication is not turned
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on or that CUCM is not accepting registrations for another reason.
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A codec mismatch means that both sides could not agree on a codec to use.
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Let’s go through these one by one and learn how to identify and fix them.
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First let’s learn how to turn on logging and find the logs on the core.
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In core manager under the Softphone tab click on ‘Edit’ and then enable logging. Then click ‘Save’.
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This will turn on logging so that we can see what’s happening on the core in real time.
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We can see messages between the core and
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the proxy including registration attempts and calls that we place.
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Then open a browser window and for the URL use the IP address of the core followed by /sip.txt.
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An example would be in the format http://192.168.0.23/sip.txt.
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This will open the SIP logging page where we can see messages between the proxy and the core.
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This log can also be used to view SIP calls as well when they are placed.
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The newest messages are at the very bottom.
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The log does not automatically slow so refresh to see new messages.
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Now that we’ve got logging turned on let’s look at some things that you might see.
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Let’s recall that in the core we had to specify a proxy IP address.
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However, sometimes we see a ‘Registration Failed
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(Service Unavailable)’ or ‘Request Timeout’ in the softphone status
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after we complete our configuration.
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There could be a couple of reasons.
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One reason might be is that the proxy is not reachable on the LAN interface we picked earlier.
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If we enable logging on the softphone and look at our sip.txt file we may see the message above.
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We see where we sent a message to the IP address of the proxy and got a ‘Request
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Timeout’ because we got no response.
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If this happens check with your network administrator to make sure the core is using
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the correct LAN interface and that you have the correct IP address for CUCM.
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Let’s go back and look at another possible issue.
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The Username in the softphone refers to an extension configured on the call manager.
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They must match or the softphone will not register.
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If ‘1001’ is not a valid extension on the call manager
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we will see a ‘Not Found’ error when the softphone tries to register.
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Here we see that message in the softphone block as our status.
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If we go back to CUCM, in this examples we see that we have a Directory Number of ‘2010’
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which does not match what we configured on the core manager.
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If we correct them so both sides match then we should be able to register.
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If we place a call we may see this message in the softphone block.
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What does this mean? Let’s take a look at the sip.txt log and see what it shows.
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The sip.txt log shows us this.
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This means that we tried to set up a call using a codec that CUCM doesn’t have enabled.
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Go back to core manager
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and try enabling all codecs if they aren’t already and try the call again.
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Also make sure that SRTP isn’t accidentally checked while UDP is being used.
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SRTP should only be checked if TLS is also checked.
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That’s it for basic SIP registration with CUCM.
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In future trainings we will discuss Advanced Registration on CUCM with multiple softphones
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and SIP trunking for CUCM.