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Replies

IPIPGW with the verizon IP trunking Service

dconstantino
Level 4
Level 4

I was wondering if anyone has so example Dial-peer configuration for doing a IPIPGW. My customer is signed up using a IP Trunking service in a MPLS cloud with VzB. I registered the 3845 router with all the proper IOS and Lic into the CCM 4.2(3). The GW is setup to do H323 to SIP see config below. I will be running 700 sites off of two clusters and 2 IP to IP GW. Sites are small.

voice-card 0

dspfarm

dsp services dspfarm

!

!

!

!

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

h323

sip

bind control source-interface GigabitEthernet0/0

bind media source-interface GigabitEthernet0/0

rel1xx disable

!

!

!

voice class codec 1

codec preference 1 g729r8

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface GigabitEthernet0/0

description IPIPGW Link to 10.x.x.0 Network Cluster A

ip address 10.x.x.254 255.255.255.0

ip virtual-reassembly

load-interval 30

duplex full

speed 1000

media-type rj45

h323-gateway voip interface

!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

media-type rj45

!

ip route 0.0.0.0 0.0.0.0 10.x.x.1

!

!

ip http server

no ip http secure-server

!

!

!

!

!

!

!

control-plane

!

!

!

voice-port 0/1/0

!

voice-port 0/1/1

!

voice-port 0/1/2

!

voice-port 0/1/3

!

!

!

sccp local GigabitEthernet0/0

sccp ccm 10.x.x.x identifier 2 priority 2 version 4.1

sccp ccm 10.x.x.x identifier 1 priority 1 version 4.1

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register DC1MTPIPIPGW

!

dspfarm profile 1 mtp

description MTP for IPIPGW Data Center 1

codec g711ulaw

maximum sessions software 250

associate application SCCP

!

!

dial-peer voice 100 voip

description voip dial peer - To Verizon IP Trunk

destination-pattern .T

rtp payload-type cisco-codec-fax-ack 114

rtp payload-type cisco-codec-fax-ind 113

voice-class codec 1

session protocol sipv2

session target sip-server

incoming called-number .

dtmf-relay rtp-nte digit-drop

!

dial-peer voice 101 voip

description voip dial peer - Inbound to CCM SUB1 CLA

preference 2

destination-pattern .T

session target ipv4:10.195.32.3

incoming called-number .

dtmf-relay h245-alphanumeric

!

dial-peer voice 102 voip

description voip dial peer - Inbound to CCM SUB2 CLB

preference 3

destination-pattern .T

session target ipv4:10.195.16.4

incoming called-number .

dtmf-relay h245-alphanumeric

!

dial-peer voice 201 voip

description voip dial peer - Inbound to CCM SUB1 CLB

preference 4

destination-pattern .T

session target ipv4:10.195.16.3

incoming called-number .

dtmf-relay h245-alphanumeric

!

dial-peer voice 202 voip

description voip dial peer - Inbound to CCM SUB2 CLA

preference 5

destination-pattern .T

session target ipv4:10.195.32.4

incoming called-number .

dtmf-relay h245-alphanumeric

!

!

gateway

timer receive-rtp 1200

!

sip-ua

retry invite 2

retry bye 2

retry cancel 2

retry options 1

sip-server dns:abcxyz.com

g729-annexb override

!

!

!

gatekeeper

shutdown

!

8 Replies 8

jposada_us
Level 1
Level 1

dconstantino, are you asking for a dial-peer example because you are geeting problems for calling to the IP Trunk or just to have an example and compare your configuration? I sew yur configuration and look fine.

I have exactly the same configuration as you have, CCM 4.1(3) with 2801 IPIPGW IOS 12.4(6)T7 and a SIP trunk to a carrier, The IPIPGW is doing interworking between H323 & SIP and it's working fine.

Maybe what is missing is the "h323-gateway voip bind srcaddr 10.x.x.254" line under the interface GigabitEthernet0/0, the "bind interface" under the sccp ccm group 1

Here is mi dial peer to the PSTN via the SIP IP trnunk to the carrier. MTP is enable for calling to PSTN

dial-peer voice 1009 voip

description dial peer to PSTN

destination-pattern .T

session protocol sipv2

session target ipv4:sip-server

dtmf-relay rtp-nte

codec g729r8 bytes 30

fax rate 9600

ip qos dscp cs3 media

ip qos dscp cs3 signaling

I hope it can help you.

go it working it was Verizon not me. Thanks anyway

So, is this really working with G.729 as your config suggests? Do you have G.711/H.323 from CM to the IPIPGW, then G.729/SIP to the carrier?

I thought that was not supported? Does it just require the latest DSPs (C5510s)?

end to end G729

No DSP required because it is just IP to IP GW

You have to have a special IOS Cisco sends you a disk and Lic.

Then you confogure Enhanced IOS MTP's in CCM

And setup the SCCP and CCM groups on your IPIPGW

What you mean when you said "No DSP requiered"?

I believe for enhaced IOS MTP you do need DSP resources.

can you confirm that please?

This is an IP to IP Gateway all it dose depending on how you set it up H323 to SIP,SIP to SIP, or H323 to H323 it is a passthourgh for the RTP stream.

The IOS MTP is software. look at my configuration.

The only reason I have DSP on this router because It use to run TDM voice (PRI) on it and I converted it to a IPIPGW.

You do not need DSP to route data traffic all you are doing is routing IP traffic and then the service provided sends it out to the PSTN.

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5640/products_configuration_guide_book09186a0080409b6d.html

Read this Doc

I have 15 pilots om two IPIPGW and starting my main rollout in Dec. 800 sites two CCM clusters.

So you are using only the CM IOS Enhanced Software MTP with this? This is all VERY interesting. I was on an engagement a while ago where the Cisco solution lost out due to lack of ability to trunk to VZB via G.729 with SIP. Only G.711 was supported. So, not it seems, G.729 is finally supported!

Your initial configuration was for G.711 MTP only, though. Could you post your final config?

I understand. you are doing passthrough for te RTP, the IPIPGW does not join any rtp traffic.

The reason i ask you is because I have a similar configuration, an ipipgw doing interworking between sip/h.323 but have to perform SBC functions since i have the customer network talking to the ISP network. so the ipipgw join all rtp streams.

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