cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
5996
Views
47
Helpful
25
Replies

Implementing V.150.1 MER on CUCM 11.5 system w/ISR4431 Gateways

RAustin70
Level 1
Level 1
 

New guy at this whole UC thing.  From May to September we installed an all new Cisco VoIP system on the site with 3k endpoints.

 

We were just told we have to now support secure VoIP using vIPr devices (cisco 7960 based devices).  So I am hitting the books, and found that we need V.150.1 MER on the 4431's, and they do meet all prereq's per the Cisco V.150.1 MER config guide.  In going through that guide I came to  the document - Configure V.150 with CUCM, and started rolling through setting up the MRG's and MRG Lists for V.150 and non-V.150 Endpoints.

 

Now, I am on to configuring the Gateway for V.150, and I am stuck.  In the document it wants me to create new gateways and set up the ports.  We already have both 4431 gateways configured as MGCP gateways with Module ISR-3NIM-MBRD, Sub 1 NIM-4MFT-T1E1-T1, and Sub 2 NIM-2MFT-T1E1-T1.  All of these ports are in use on each device (1 DSN, 2 Local, 1 LD, & 2 back to the old PBX) running Digital Access PRI Protocol with a Media Resource Group List of SITE-MAIN.

 

The (first) question I have is - Does this guide intend me to modify existing gateways, or create new gateways on the same hardware?  Feeling way out of my element on this one.

25 Replies 25

Hope I can choose either protocols sip or sccp right?

From what I understand as of October last year(last time I checked), only SCCP was approved for use in the DoD arena, but SIP was forthcoming upon approval.  I just checked the latest JITC DTR dated 14 July 2023, and it lists:

Analog PSTN mode DSCD GD vIPer for Env 1, 2, 3; versions 6.1.2.1/6.1.2.2
SCCP and SIP modes DSCD GD IP vIPer for Env 1, 2, 3; versions 6.1.2.1/6.1.2.2

Both came with a sub-note of "Although the SUT was tested and is certified with this GD vIPer DSCD version, JITC analysis determined the SUT is also certified with other versions previously and currently listed on the DoDIN APL as denoted under a separate tracking number for the GD vIPer DSCD."

 

So long story short, looks like SIP is good to go, but I am still SCCP until I get a chance to play with SIP in a testbed.

I was able to change the registration from third party sip phone to sccp which was easy. Now, when the owner of that phone tries to re-key the phone by dialing a number in NSA, I hear a modem tone and call just drops. I don't understand anything about the re-keying stuffs so I am totally lost here. I have enabled V.150 as well as "include V.150 MER Capabilities". Am I missing anything?

If you are on a DoD site, you may have VPS running.  that monitors for fax/modem energy and if it hears it, checks a whitelist to see if that DN is allowed to send data.  if not, it will drop the call maybe 5 seconds into hearing the fax or modem tones.  That would be my best guess.  I also do not know anything about the rekey, that is COMSEC office responsibilities not UC team.

We are not on the DoD site nor the number blacklisted. When I heard him re-keying the phone from other side of the cube, I heard the modem tone but for some reason the handshaking couldn't take place and the call dropped in a few seconds. I don't know anything about the re-keying but I just wanted to make sure the phone is good from the configuration point of view.

Ours are SCCP, the latest version does include a soft modem so v.150 is no longer used.

You can leave the v.150 filter set for Default, no changes are needed. We used 7960 but other models probably are supported. see attached screenshots for settings on the phone software GUI.Capture 1.PNGCapture 2.PNGCapture 3.PNGCapture 4.PNGCapture 5.PNG

I agree with Charles, has to be VoIP, not using the PSTN Connect dongle.  also, if you load version 6.1.2.2 it comes with a softmodem that takes the place of V.150MER if it does not detect it operating on the network.  I wish they had that when I started this post TBH.  Made life much easier.

I am trying to support v.150 for viper analog phones over a sip trunk to a 4431 router sip-sip. I see a conf guide for cucm but i don't see a config guide for the router itself,

heathmiller75
Level 1
Level 1

My main issue with Vipers and or STE's transmitting or rekeying over a PRI T-1(MGCP) is to ensure your timing is rock solid. If you have frame slips across your PRI's the calls with drop when secure. Thanks for the additonal setting for the VIper up above with key an out out for issues once i roll over to SIP trunks.

TournITquets
Level 1
Level 1

To get V.150 to passthrough a CUBE setup, make sure that you configure SDP passthrough on the VG according to https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_sip/configuration/xe-3s/cube-sip-xe-3s-book/m_voi-cube-v150-mer-sdp-passthru.htmlSDP passthrough mode "is added in CUBE (IP-to-IP gateway) to support the V.150.1 MER modem relay in SDP passthrough mode. This is of importance when CUBE is in the media path between V.150.1 MER supported gateways and needs to handle non-RTP packets (such as SPRT packets)".  Let me know if this works.