SBC Compatibility with Cisco MGCP Gateway for Call Processing

[min ko]

Dear Experts,
I am reaching out to seek your expertise on a significant transition we are planning within our telecommunications infrastructure.

Currently, we are utilizing a CUCM on-premise solution with PSTN Cisco MGCP routers. The existing architecture involves an ITSP Provider connecting to our Cisco T1/MGCP Gateway, which, in turn, interfaces with CUCM using the MGCP protocol.

Due to evolving business needs, we are in the process of migrating to a Cloud MS PBX solution, necessitating a change in our overall system design. While CUCM, UCCX, and IMP will be phased out, we intend to retain the MGCP protocol at our Cisco VG Gateway. This decision is based on the unavailability of SIP trunking services from our local ITSP.

The proposed system design involves introducing a new third-party SBC between the on premise ITSP (T1 analog Provider) and new Microsoft PABX at Azure Cloud.

Current Connection:
ITSP Provider > Cisco T1/MGCP Gateway > CUCM (MGCP)

Proposed System Design:
ITSP Provider <> Cisco (MGCP/SIP) <> New Third-Party SBC

To facilitate this transition, we plan to purchase a CUBE license to do sip  to sip trunking on the router for internal call legs. External ITSP call legs will be maintained using T1/RJ11 ports configured for MGCP.

Our primary concern is ensuring compatibility between the Cisco MGCP Gateway and the new SBC. Specifically, we seek your insights into whether the SBC can effectively understand the MGCP protocol employed by the Cisco VG Gateway after SIP trunking is established between them like CUCM server. Our objective is to ensure seamless call processing working or not for both incoming and outgoing phone calls.

Any guidance, recommendations, or considerations you can provide regarding this migration would be immensely valuable. We appreciate your time and expertise in advance.

Thank you and best regards,

Min

[Maren Mahoney]

The MGCP Protocol requires a Call Agent such as CUCM. It is unlikely that your SBC can act as a call agent, but the SBC organization will know.

A better solution might be fully terminating the ISDN circuit on your gateway (rather than using MGCP) and then using a SIP Trunk from your gateway to the SBC. No MGCP, no CUCM, no Call Agent required. And there are any number of docs you can search for that will give you insight on how to configure the ISDN side, the SIP side, and the integration between the two on your gateway.

Maren

[min ko]

Dear Maren,

Thanks for your quick response.

So , do you mean our new Proposed System Design will not working call processing (inbound, outbound call) well. correct?

Current Connection:
ITSP Provider > Cisco T1/MGCP Gateway > CUCM (MGCP)

Proposed System Design:
ITSP Provider <> Cisco (MGCP/SIP) <> New Third-Party SBC <> Cloud PBX

In our country, Sip trunking and ISDN circuit services are not currently available. 

May I know that how to implementing a full termination setup on our Cisco gateway without using MGCP. what protocol do we need in cisco router?

We would greatly appreciate it if you could provide us with more detailed information on the steps and configurations required for this setup.

In the context of establishing a SIP-to-SIP (internal call leg) connection between our Cisco VG gateway and a new SBC for the ingress call leg, we would like to confirm whether it is necessary to enable the CUBE (Cisco Unified Border Element) function on the Cisco router.?

Additionally, Do we need to purchase a specific CUBE license from Cisco to enable this functionality? If I am mistaken or your correction and guidance on the matter would be immensely helpful.

Our customer has carefully evaluated their requirements and options and has chosen a new SBC solution that aligns with their business needs. @Nithin Eluvathingal 

Your expertise in this area is highly valued, and I appreciate your time and assistance in clarifying these technical aspects.

Best regards,

Min

[Roger Kallberg]

With MS Teams calling you would have this topology.

Service Provider (TDM or SIP trunk) -> Cisco router, acting as a SBC, ie a SIP trunk from your service provider, or a traditional voice gateway, ie an ISDN or other TDM voice service from your service provider -> MS Teams

[Maren Mahoney]

So you have a channelized T1 emulating 24 analog ports, rather than a PRI. I misread your original post.

Like a PRI, you can terminate the Channelized-T1 on the gateway and use SIP to connect to the SBC. You can configure the T1 controller with a 'ds0-group' which will create voice port you can reference in dial-peers on the ITSP side. You'd have another set of dial-peers configured as SIP voip dial-peers on the SBC side.

You would not need a CUBE license on the Analog-T1 to SIP gateway. A CUBE license is needed when interconnecting two VoIP connections. (For example: If you had a SIP Trunk from your ITSP to the GW and then another SIP Trunk from the GW to your SBC, to get the most functionality you'd need a CUBE license.)

It's been many years since I have configured something like this and the router platforms have changed quite a bit. What specific router/gateway do you have and what IOS is it running? Which kind of VWIC or NIM is anchoring your T1?

Maren

[min ko]

Dear Maren,

Thanks much for your reply.

We have not Centralized T1 (ISDN) with 24 analog ports We have 16 x FXO/FXS (PSTN) ports land line. one RJ 11 port with one DID. number. you may check as following config. I look forward your reply. And do let me know new propose design will working or not , ASAP. thanks much again.

version 15.3
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
service password-encryption
!
hostname VGW02
!
boot-start-marker
boot-end-marker
!
aqm-register-fnf
!
logging buffered 10000
!
no aaa new-model
!
!
ip traffic-export profile sniffer mode capture
bidirectional
!
!
!
!
!
!
ip host CUCM-01.brilliance.com 192.168.1.10
ip host CUCM-02.brilliance.com 192.168.1.11
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
!
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
!
voice class dualtone-detect-params 100
freq-max-deviation 50
freq-min-power 10
!
voice class custom-cptone test
dualtone busy
frequency 450
dualtone ringback
frequency 400
dualtone reorder
frequency 480
cadence 480 480
dualtone disconnect
frequency 400 450
cadence 500 500
!
voice class custom-cptone Disconnect
dualtone disconnect
frequency 425
cadence 250 250
!
!
!
!
!
!
license udi pid CISCO2911/K9 sn FGL190410RX
hw-module pvdm 0/0
!
!
!
!
redundancy
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address 172.16.1.121 255.255.255.0
ip traffic-export apply sniffer size 20000000
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.1.121
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 172.16.1.1
!
kron occurrence SHUT_FXO1 at 4:00 recurring
policy-list SHUT_NO_SHUT_POL1
!
kron policy-list SHUT_NO_SHUT_POL1
cli event manager run SHUT_NO_SHUT1
cli event manager run SHUT_NO_SHUT2
cli event manager run SHUT_NO_SHUT3
!
!
!
snmp-server community WANMonitor RO
!
control-plane
!
!
voice-port 0/0/0
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653239"
!
voice-port 0/0/1
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing sup-disconnect 100
description "CO line number 653240"
!
voice-port 0/0/2
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653241"
!
voice-port 0/0/3
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653242"
!
voice-port 0/1/0
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653243"
!
voice-port 0/1/1
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653245"
caller-id enable
!
voice-port 0/1/2
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653246"
caller-id enable
!
voice-port 0/1/3
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653247"
caller-id enable
!
voice-port 0/2/0
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653248"
!
voice-port 0/2/1
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 3
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 653249"
!
voice-port 0/2/2
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "SG CO line number 62387753"
!
voice-port 0/2/3
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "SG CO line number 62386539"
!
voice-port 0/3/0
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 651305"
!
voice-port 0/3/1
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 651840"
!
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 656980"
!
voice-port 0/3/3
supervisory disconnect dualtone mid-call
no battery-reversal
disc_pi_off
timeouts initial 5
timeouts interdigit 2
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
timing hookflash-out 50
timing guard-out 1000
timing sup-disconnect 100
description "CO line number 651902"
!
!
!
!
!
ccm-manager redundant-host CUCM-02.daewooenp.com
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server CUCM-01.daewooenp.com CUCM-02.daewooenp.com
ccm-manager config
!
mgcp
mgcp call-agent CUCM-01.daewooenp.com 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp bind control source-interface GigabitEthernet0/0
mgcp bind media source-interface GigabitEthernet0/0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 172.16.1.110 identifier 2 version 7.0
sccp ccm 172.16.1.109 identifier 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 1 register conferrence
associate profile 2 register MTP_mtp
associate profile 3 register trancode
!
dspfarm profile 3 transcode
description ***** Trancoder *****
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 5
associate application SCCP
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
maximum sessions 3
associate application SCCP
!
dspfarm profile 2 mtp
description ***** Media Termination Point *****
codec g711ulaw
codec pass-through
maximum sessions software 500
associate application SCCP
!
dial-peer voice 99900990 pots
description FXO
service mgcpapp
port 0/1/2
!
dial-peer voice 99900991 pots
description FXO
service mgcpapp
port 0/0/1
!
dial-peer voice 99900992 pots
description FXO
service mgcpapp
port 0/0/2
!
dial-peer voice 99900993 pots
description FXO
service mgcpapp
port 0/0/3
!
dial-peer voice 99901990 pots
description FXO
service mgcpapp
port 0/1/0
!
dial-peer voice 99901991 pots
description FXO
service mgcpapp
port 0/1/1
!
dial-peer voice 99901993 pots
description FXO
service mgcpapp
port 0/1/3
!
dial-peer voice 99902990 pots
description FXO
service mgcpapp
port 0/2/0
!
dial-peer voice 99902991 pots
description FXO
service mgcpapp
port 0/2/1
!
dial-peer voice 99902992 pots
description FXO
service mgcpapp
port 0/2/2
!
dial-peer voice 99902993 pots
description FXO
service mgcpapp
port 0/2/3
!
dial-peer voice 99900099 pots
service mgcpapp
port 0/0/0
!
dial-peer voice 99900199 pots
service mgcpapp
port 0/0/1
!
dial-peer voice 99900299 pots
service mgcpapp
port 0/0/2
!
dial-peer voice 99900399 pots
service mgcpapp
port 0/0/3
!
dial-peer voice 99901099 pots
service mgcpapp
port 0/1/0
!
dial-peer voice 99901199 pots
service mgcpapp
port 0/1/1
!
dial-peer voice 99901992 pots
service mgcpapp
port 0/1/2
!
dial-peer voice 99901399 pots
service mgcpapp
port 0/1/3
!
dial-peer voice 99902099 pots
service mgcpapp
port 0/2/0
!
dial-peer voice 99902199 pots
service mgcpapp
port 0/2/1
!
dial-peer voice 99902299 pots
service mgcpapp
port 0/2/2
!
dial-peer voice 99902399 pots
service mgcpapp
port 0/2/3
!
dial-peer voice 99901299 pots
service mgcpapp
port 0/1/2
!
dial-peer voice 99903990 pots
service mgcpapp
port 0/3/0
!
dial-peer voice 99903991 pots
service mgcpapp
port 0/3/1
!
dial-peer voice 99903992 pots
service mgcpapp
port 0/3/2
!
dial-peer voice 99903993 pots
service mgcpapp
port 0/3/3
!
dial-peer voice 99903099 pots
service mgcpapp
port 0/3/0
!
dial-peer voice 99903199 pots
service mgcpapp
port 0/3/1
!
dial-peer voice 99903299 pots
service mgcpapp
port 0/3/2
!
dial-peer voice 99903399 pots
service mgcpapp
port 0/3/3
!
!
gateway
timer receive-rtp 600
!
!
!
gatekeeper
shutdown
!
!
!
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
login local
length 0
transport input all
!
scheduler allocate 20000 1000
ntp source GigabitEthernet0/0
ntp master 4
ntp server 172.16.1.121
event manager applet SHUT_NO_SHUT1
event none
action 1.0 cli command "enable"
action 1.1 cli command "conf t"
action 2.0 cli command "voice-port 0/0/0"
action 2.1 cli command "shut"
action 2.2 wait 2
action 2.3 cli command "no shut"
action 3.0 cli command "voice-port 0/0/1"
action 3.1 cli command "shut"
action 3.2 wait 2
action 3.3 cli command "no shut"
action 4.0 cli command "voice-port 0/0/2"
action 4.1 cli command "shut"
action 4.2 wait 2
action 4.3 cli command "no shut"
action 5.0 cli command "voice-port 0/0/3"
action 5.1 cli command "shut"
action 5.2 wait 2
action 5.3 cli command "no shut"
event manager applet SHUT_NO_SHUT2
event none
action 1.0 cli command "enable"
action 1.1 cli command "conf t"
action 2.0 cli command "voice-port 0/1/0"
action 2.1 cli command "shut"
action 2.2 wait 2
action 2.3 cli command "no shut"
action 3.0 cli command "voice-port 0/1/1"
action 3.1 cli command "shut"
action 3.2 wait 2
action 3.3 cli command "no shut"
action 4.0 cli command "voice-port 0/1/2"
action 4.1 cli command "shut"
action 4.2 wait 2
action 4.3 cli command "no shut"
action 5.0 cli command "voice-port 0/1/3"
action 5.1 cli command "shut"
action 5.2 wait 2
action 5.3 cli command "no shut"
action 6.0 cli command "voice-port 0/2/0"
action 6.1 cli command "shut"
action 6.2 wait 2
action 6.3 cli command "no shut"
action 7.0 cli command "voice-port 0/2/1"
action 7.1 cli command "shut"
action 7.2 wait 2
action 7.3 cli command "no shut"
event manager applet SHUT_NO_SHUT3
event none
action 1.0 cli command "enable"
action 1.1 cli command "conf t"
action 2.0 cli command "voice-port 0/2/2"
action 2.1 cli command "shut"
action 2.2 wait 2
action 2.3 cli command "no shut"
action 3.0 cli command "voice-port 0/2/3"
action 3.1 cli command "shut"
action 3.2 wait 2
action 3.3 cli command "no shut"
action 4.0 cli command "voice-port 0/3/0"
action 4.1 cli command "shut"
action 4.2 wait 2
action 4.3 cli command "no shut"
action 5.0 cli command "voice-port 0/3/1"
action 5.1 cli command "shut"
action 5.2 wait 2
action 5.3 cli command "no shut"
action 6.0 cli command "voice-port 0/3/2"
action 6.1 cli command "shut"
action 6.2 wait 2
action 6.3 cli command "no shut"
action 7.0 cli command "voice-port 0/3/3"
action 7.1 cli command "shut"
action 7.2 wait 2
action 7.3 cli command "no shut"
!
end

Best rgds,

Min

[Roger Kallberg · Accepted solution]

As me, @Maren Mahoney and @Nithin Eluvathingal already answered you MGCP will not work without CM, so no you’re suggested design will not work. Regardless of what type of connection you have with your PSTN service provider you’ll need to reconfigure your gateway to use another control protocol than MGCP. The recommendation for this would be to use SIP as that makes the setup following current best practices. Again using SIP from your gateway to the calling platform has nothing whatsoever to do with what connection type you have with your PSTN service provider.

[min ko]

Dear Roger,

Thanks a lot for your kind help, your answer is very helpful to us. Thank you  all experts.

Sincerely,

Min

[min ko]

Dear Roger,

let's assume as the new scenario where we transition from MGCP to SIP and establish a SIP trunk connection between the Cisco Cube and the new SBC,

*PSTN analog provider (RJ-11) pots > cisco cube (sip) > new SBC (sip)

As I understand, the operational processes of IP PBX and SBC differ significantly.

I am seeking guidance on how the new SBC will see the PSTN physical voice POTS (analog) inside the Cisco router, similar to the way CUCM IP PBX server handles it.

I would greatly appreciate your insights and guidance on this matter.

Thank you for your time and expertise.

[Roger Kallberg]

I can’t tell you how this operates in the other SBC, but on the Cisco SBC (Cube) side it is handled by dial peers. In one of the documents I shared earlier you can find how that operates.

[Nithin Eluvathingal]

As @Maren Mahoney explained, MGCP is a Call Agent/Endpoint protocol, where the Endpoint is controlled by  CUCM. The CUCM has the entire control intelligence and instructs the Endpoint what action to take once an event is detected.

Removing the service MGCP from the controller will make the gateway control the PRI. And using a SIP dial-peer, you can send calls to the SIP trunk you establish between the gateway and MS Phone System.

If you are using ISR 4K or above, you don’t need to add a separate SBC, as Cisco ISR 4K can be used for direct routing. Here is the link."

: https://www.cisco.com/c/en/us/solutions/collaboration/microsoft-teams-direct-routing.html

[Roger Kallberg]

I concur with @Maren Mahoney. It is highly unlikely that the SBC would support MGCP. Apart from that there seems like there might be a slight possibility that you have a misunderstanding on how the different call processing protocols operates. You can use SIP as the control protocol between your gateway and CM even if you don’t have a SIP trunk to your service provider. Using SIP as the control protocol between gateways and CM regardless of the interface with the service provider is pretty much the de-facto standard these days.

[soniyadixit]

Okay, think of it like this: if you have a special phone system (SBC) and a particular type of gateway (Cisco MGCP), you want to make sure they talk nicely to each other. It's like checking if your English-speaking friend understands your Spanish. You're just making sure they can work together smoothly to handle phone calls.

[min ko]

Hi Soniya,

Thanks for acknowledging.
Now, I'm figuring out how to established SIP trunking from Cisco (CUBE) router to a non-Cisco SBC since we don't have CUCM anymore.

As I know ,sip trunking can do in CUCM.And CUBE can do SIP-UA registration, but I'm not sure if it's the same as SIP trunking in Cisco CUBE like CUCM.

If not, could you please provide docs or guide on how to establish SIP trunking from our Cisco SIP (CUBE) router to a non-Cisco SBC?

Your expertise on this matter would be highly valuable, and I look forward to your guidance.

Looking forward to your response.

best rgds,

Min

[Roger Kallberg]

It’s not anything special to setup a trunk between a Cisco Cube and another SBC. Have a look at the document for a general idea of how to do this. Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 

Also please have a look at this document for how call routing works in IOS. Explain Cisco IOS and IOS XE Call Routing

If you need to use SIP profile(s) to get the connection operational this document should provide you with a starting point for that. Use SIP Profiles on CUBE Enterprise Common Use Cases