TA 900/900e as a MGCP VOIP/IAD as opposed to Voice User SIP Lines. What are the disadvantages?

[Anonymous]

Does anyone know what the advantages or disadvantages are to delivering analog lines through FXS ports on a TA 900/900e as a MGCP VOIP/IAD.  Is this as secure as Voice User and SIP?

[Anonymous · Accepted solution]

The advantages & disadvantages really depend on the feature & services offered by the ITSP. That being said, an ITSP looking for more flexibility, feature support, and call control options "in the gateway"; then SIP
is the way to go. However, if they prefer less flexibility in the gateway, in which all Intelligence, call control and features are handled by the upstream Softswitch/Feature server; then MGCP is the better option. They should also consider the technical depth of their staff, as SIP configuration in the gateway is more complex than MGCP.

In regards to security, MGCP is more secure in this type of deployment. By moving intelligence to the gateway/customer premise (with SIP); there's greater potential for sniffing/capturing information from the SIP/SDP; not to mention hijacking the device and placing calls. With MGCP, less harmful/sensitive information is sent with the call control, and Hijacking is virtually impossible (since it only communicates with configured Call Agents).

[Anonymous]

That is exactly what I was thinking... MGCP is great for provisioning folks because the majority of our Phone service is currently rolled out on EMTAs.  We also like the way it fits into the Databases and billing system on the periphery.  Security is more of a concern for the OffNet subscriber so we will be able to force Authentication through our softswitch Remote Media Gateway models, and can check off that Box.

The real worry now is the credit card machines, alarms systems and fax machines.  That may be problematic.  Have you tested any of these?

[Anonymous]

I have tested all three and I have found Adtran to be really good with these.

• Credit card - never had a problem
• Fax - 95% or better
• Alarms - Usually not a problem but it depends on the central station

Make sure you have the configuration properly set for data/fax transmission.

Ultimately it will depend on the quality of the connection.

[Anonymous]

That sounds good...  do you find that you still need to add modem passthrough for all these instances?  T38 for all fax lines, or do you let it fly with G711 only?

[Anonymous]

I have it configured for both modem-passthrough and t38.

At times I do have to tweak the t38 settings for faxing but usually the default settings work fine.

Here is a document that has more info on the t38 options:

https://supportforums.adtran.com/servlet/JiveServlet/previewBody/1827-102-3-3616/T.38%20Protocol%20i...

See the bottom of page 8 for the standard recommended settings.

[Anonymous]

We also have done a variety of T38 configurations for all our SIP/FXS handoffs...  so far my testing with this MGCP configuration has made me lean more toward no T38 settings at all.

[Anonymous]

G711 will work fine as long as you don't have packet loss or too much network jitter.

How are you connecting to the PSTN?

[Anonymous]

Well that is the rub...  We connect via TDM and SIP Trunks.  We are always attempting to use the TDM path and keep the routes from creating too many variables from entering the equation.

[Anonymous]

If you use SIP trunks over the public internet I highly recommend you use t38 for faxing.

Unless you have a really good connection to the internet you will run into problems.

[Anonymous]

That's sounds good.  I rarely trust network elements we have no or little control over.  When you are deploying a T1/PRI circuit and have a SIP Binding, do you also build a stand alone SIP trunk for the MGCP traffic usage?  or do you let them all blend together?

[Anonymous]

It really depends on the setup.

What I have done when there is issues with data transmissions is that I route those calls (either by ANI or steering digit) to TDM or POTS lines and the voice I send over the public internet to a SIP trunk.

[Anonymous]

that sounds like what I want to start doing...  I really am uncertain of the best way to do that right now with our distributed routing in many different NPAs and LATAs.   Our best case scenario is long distance calling which I can point to TDM specific trunks more easily than the local call.  The real problematic calls are the where the subscriber is trying to mix use a line for Credit Card, Voice, Alarm or FAX.

How would you use ANI or Steering Digit?

[Anonymous]

Take a look at this document on SABR:

https://supportforums.adtran.com/servlet/JiveServlet/previewBody/1862-102-2-1978/Source%20and%20ANI%...

You can create an ANI list and then route it to a specific trunk based upon it.

To use steering digits you will need to prepend digits on to the number dialed.

You will than need to create a trunk that will only accept calls with those specific digits prepended. (and strip off the prepended digits before sending out the call)

I believe in most situations SABR will be easier to configure.

[Anonymous]

That sounds great...  i will tinker with that also and see if I can work it out.

When I am deploying a T1/PRI circuit and have a SIP Binding, is it a good idea to also build a stand alone SIP trunk for the MGCP traffic usage?  or should I let them all blend together?

[Anonymous]

I see no reason to unless you are experiencing issues and need to debug.

By separating the traffic you will have an easier time sorting through the debug messages.

[Anonymous]

rthompson,

I went ahead and flagged this post as "Assumed Answered".  If any of the responses on this thread assisted you, please mark them as Correct or Helpful as the case may be with the applicable buttons.  This will make them visible and help other members of the community find solutions more easily.  If you still need assistance, I would be more than happy to continue working with you on this - just let me know in a reply.

Thanks!

David

[Anonymous]

Rthompson,

I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it and select another in its place with the applicable buttons.  If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

Thanks,

David