cisco voip design fundamentals

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CHAPTER BETA DRAFT - CISCO CONFIDENTIAL -- 1/22/02 2-1 Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide 2 Voice Networks: Design Fundamentals Introduction Voice over IP (VoIP) networks rely upon the H.323 standard for the transmission of real-time audio communications over packet-based networks. The Cisco SS7 Interconnect for Voice Gateways 2.0 solution enables a VoIP network to interconnect with an SS7-based TDM network. This chapter covers the fundamental aspects essential to voice networks in a Cisco SS7 Interconnect for Voice Gateways 2.0, and presents the following major topics: Designing and Provisioning H.323 VoIP Networks Using a Remote Cisco SLT Using the Generic Transparency Descriptor for GKTMP Gateway Configuration Examples Configuring a Voice Gateway for Universal Service Configuring a Gatekeeper Configuring TDM Switching Services Managing Echo Cancellation Note This chapter is intended to provide a background of the fundamentals of VoIP networks. As such it may discuss features and topics that are not relevant to your network. Note The sample configurations that follow may include settings and parameters that are not applicable to your network. Be sure to substitute values where appropriate to accommodate the needs of your network. Designing and Provisioning H.323 VoIP Networks The fundamentals of designing and provisioning H.323 networks for VoIP services are documented at the following location: http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/multi_c/mcprt1/mcdvoip.ht m

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Cisco VOIP Design Fundamentals

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Page 1: Cisco VOIP Design Fundamentals

C H A P T E R

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2-1Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

2Voice Networks: Design Fundamentals

IntroductionVoice over IP (VoIP) networks rely upon the H.323 standard for the transmission of real-time audiocommunications over packet-based networks. The Cisco SS7 Interconnect for Voice Gateways 2.0solution enables a VoIP network to interconnect with an SS7-based TDM network.

This chapter covers the fundamental aspects essential to voice networks in a Cisco SS7 Interconnect forVoice Gateways 2.0, and presents the following major topics:

• Designing and Provisioning H.323 VoIP Networks

• Using a Remote Cisco SLT

• Using the Generic Transparency Descriptor for GKTMP

• Gateway Configuration Examples

– Configuring a Voice Gateway for Universal Service

– Configuring a Gatekeeper

– Configuring TDM Switching Services

• Managing Echo Cancellation

Note This chapter is intended to provide a background of the fundamentals of VoIP networks. As such itmay discuss features and topics that are not relevant to your network.

Note The sample configurations that follow may include settings and parameters that are not applicable toyour network. Be sure to substitute values where appropriate to accommodate the needs of yournetwork.

Designing and Provisioning H.323 VoIP NetworksThe fundamentals of designing and provisioning H.323 networks for VoIP services are documented atthe following location:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/multi_c/mcprt1/mcdvoip.htm

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2-2Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsUsing a Remote Cisco SLT

Using a Remote Cisco SLTBeginning with Release 2.0 of the Cisco SS7 Interconnect for Voice Gateways Solution, Cisco supportsnetwork configurations in which the Cisco SLT is remotely located from the Cisco PGW 2200. However,in order to ensure an adequate level of service, the network must be configured to meet the followingconditions:

Note These recommendations do not guarantee 100% call completion rates or uninterrupted service fromSS7 links. To ensure the highest levels of service, Cisco continues to recommend that the Cisco SLTbe colocated with the Cisco PGW 2200.

• One-way end-to-end delay between the Cisco SLT and the Cisco PGW 2200 must not exceed150 ms.

• Packet loss should be below 0.5% and must not exceed 1%. If packet loss exceeds 0.5%, increasethe RUDP receive window as follows:

Step 1 Increase the size of the RUDP receive window to 64 on the Cisco SLT.

ss7 session-0 m_rcvnum 64

Step 2 On the Cisco PGW 2200, edit the/opt/CiscoMGC/etc/properties.dat file as follows:

Change*.rudpWindowSz = 32 to *.rudpWindowSz = 64

Using the Generic Transparency Descriptor for GKTMPThe Generic Transparency Descriptor (GTD) for Gatekeeper Transaction Message Protocol (GKTMP)feature provides additional functionality to voice gateways and gatekeepers in a Cisco SS7 Interconnectfor Voice Gateways Solution. The generic transparency descriptor or generic telephony descriptor(GTD) format is defined in the a Cisco proprietary draft. GTD format defines parameters and messagesof existing SS7 ISUP protocols in text format and allows SS7 messages to be carried as a payload in theH.225 registration, admission, and status (RAS) messages between the GW and GK. GTD messages canalso be transported between GWs and GKs in H.323 messages. With the GTD feature, the GK extractsthe GTD message and the external route server derives routing and accounting information based uponthe GTD information provided from the Cisco Gatekeeper Transaction Message Protocol (GKTMP).

Currently routing on Cisco GWs is based on generic parameters such as originating number, destinationnumber, and port source. Adding support for SS7 ISUP messages allows the VoIP network to useadditional routing enhancements found in traditional TDM switches.

For detailed instructions on configuring GTD for GKTMP, refer to documentation at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xu/122xu2/ftgtdpay.htm

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2-3Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsUsing Gateway Trunk and Carrier Based Routing Enhancements

Using Gateway Trunk and Carrier Based Routing EnhancementsVoice wholesalers use multiple ingress and egress carriers to route traffic. A call coming in to a gatewayon a particular ingress carrier must be routed to an appropriate egress carrier. As networks grow andbecome more complicated, the dial plans needed to route the carrier traffic efficiently become morecomplex and the need for carrier sensitive routing (CSR) increases. The Gateway Trunk and CarrierBased Routing Enhancements feature implements CSR for Cisco voice gateways. Gateway featureenhancements add the following routing features:

• Implementation of trunk groups and enhanced key matches on several platforms and interfaces

• Reduction of the number of dial peers in a dial plan by using profile aggregation and multiple trunkgroup supports

• Enhanced hunting schemes

• Carrier ID support

• Trunk group label support

• Number translation profiles per trunk group, source IP group, voice port, and dial peer

• Dial peer support of multiple trunk groups with translations per trunk group

• ENUM support

• Source IP groups

• Voice over IP (VoIP) access list control

• Enhanced translation rules in SED (stream editor) regular expressions

• Incoming call blocking

• Cisco IVR 2.0 support for carrier ID based dial peer matching, incoming call blocking, and dial peernumber translation

• Call detail record (CDR) support

• Virtual private network (VPN) source routing (also referred to as static or basic carrier routing)

For detailed instructions on configuring gateway trunk and carrier based routing enhancements, refer todocumentation at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122x/122xu/122xu2/ftpg_str.htm

Gateway Configuration ExamplesThe following configuration examples are presented, with commentary:

• Configuring a Voice Gateway for Universal Service

• Configuring a Gatekeeper

• Configuring TDM Switching Services

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2-4Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

Configuring a Voice Gateway for Universal ServiceThe following examples illustrate the configuration of a Cisco media gateway that is providing voice,prepaid VoIP, and T.38 fax relay services, in addition to using CAC features. The order of presentationof the following examples is that of their appearance in the configuration file. Not all steps are requiredin all networks.

Note The Cisco SS7 Interconnect for Voice Gateways Solution is

The following configuration examples are presented:

• Defining a MIB

• Assigning Controllers and NFAS Groups

• Enabling Accounting

• Creating a Loopback Interface

• Configuring H.323 Registration

• Configuring NTP

• Assigning TACACS+ Servers

• Enabling SNMP

• Assigning RLM Groups

• Assigning Multiple RLM Groups

• Assigning RADIUS Server Hosts and Ports

• Enabling Call Treatment and IVR

• Assigning Dial Peers to Voice Ports

• Enabling RAI

Defining a MIB

Management Information Bases, or MIBs, can be used to manage data for a variety of purposes.

Step 1 Evaluate your MIB requirements.

Step 2 Set MIB parameter options. The following example applies to ISDN service.

call-history-mib retain-timer 60call-history-mib max-size 500!

Tip For a discussion of the command options retain-timer and max-size, refer to Call Detail Records(CDR) at the following URL:http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120t/120t2/cdrfm.htm

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2-5Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

Assigning Controllers and NFAS Groups

The following assigns T3 and T1 controllers and NFAS groups.

Step 1 Assign a T3 controller.

Note Refer to Chapter 3, “Basic Configuration Using the Command-Line Interface,” ofCisco AS5350 andCisco AS5400 Universal Gateway Software Configuration Guide, at the following URL:http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5350/sw_conf/53swcg/

controller T3 1/0 framing m23 clock source line cablelength 100t1 1-28 controller

Step 2 Assign T1 controllers and NFAS groups. Multiple spans (controllers) will use the same D-channel. TheD-channel configuration is on interface serial 1/0:1:23. See alsoConfiguring TDM Switching Services,page 2-12.

Note Refer to Chapter 3, “Basic Configuration Using the Command-Line Interface,” ofCisco AS5350 andCisco AS5400 Universal Gateway Software Configuration Guide, at the following URL:http://www.cisco.com/univercd/cc/td/doc/product/access/acs_serv/as5350/sw_conf/53swcg/

Refer also to the following in that chapter: Configuring ISDN PRI, Configuring the D Channels forISDN Signaling, and Configuring ISDN NFAS on CT1 PRI Groups.

controller T1 1/0:1 framing esf pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 0!controller T1 1/0:2 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 1 nfas_group 0!controller T1 1/0:3 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 2 nfas_group 0!controller T1 1/0:4 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 3 nfas_group 0!

<---snip--->

controller T1 1/0:25 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 24 nfas_group 0!controller T1 1/0:26 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 25 nfas_group 0!controller T1 1/0:27 framing esf

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Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

pri-group timeslots 1-24 nfas_d none nfas_int 26 nfas_group 0!controller T1 1/0:28 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 27 nfas_group 0

Enabling Accounting

Step 1 Enable H.323-based gateway accounting. Thevsa (vendor-specific attributes) command option isrequired to support prepaid voice services only.

gw-accounting h323 vsa <---required for prepaid service onlygw-accounting voip

Creating a Loopback Interface

Step 1 Create a loopback interface, ensuring that traffic is directed to the server in case another interface is lost.

interface Loopback0 ip address 10.44.4.4 255.255.255.0

Note Before performing this step, be sure to check for existing loopback interfaces with the showinterface loopback command.

Configuring H.323 Registration

Step 1 Configure the gateway to register with the GK.

interface FastEthernet0/0 ip address 10.40.4.4 255.255.0.0 no ip directed-broadcast duplex full speed 100 h323-gateway voip interface h323-gateway voip id z1-gk1 ipaddr 10.40.7.50 1718 h323-gateway voip h323-id z1-gw3 h323-gateway voip tech-prefix 1#

Configuring NTP

Step 1 Use the commandntp broadcast client to cause this gateway to function as the NTP (Network TimingProtocol) client, and listen to NTP broadcasts from the NTP server to synchronize the system clock. SeeConfigure Network Timing, page 4-23.

interface FastEthernet0/1

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Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

ip address 10.41.4.4 255.255.0.0 no ip directed-broadcast duplex full speed 100 ntp broadcast client <---listen for NTP server broadcasts

Assigning TACACS+ Servers

Step 1 Assign TACACS+ servers and other parameters. TACACS+ provides detailed accounting informationand flexible administrative control over authentication and authorization processes. TACACS+ isfacilitated through AAA and can be enabled only through AAA commands.

Note Refer also to Configuring TACACS+ at the following URL:http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/secur_c/scprt2/scdtplus.htm

tacacs-server host 10.100.20.20tacacs-server host 10.100.30.30tacacs-server host 10.101.30.30tacacs-server timeout 3 <---recommended; see Caution belowtacacs-server key ciscotacacs-server administration

Enabling SNMP

Step 1 Enable SNMP parameters and traps. SeeUsing SNMP, page 3-18. The following basic template isrecommended.

Note Refer also to Configuring SNMP Support at the following URL:http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/ffun_c/fcfprt3/fcf014.htm

snmp-server community public RWsnmp-server enable traps snmp authentication linkdown linkup coldstart warmstartsnmp-server enable traps fru-ctrlsnmp-server enable traps entitysnmp-server enable traps envmonsnmp-server host 10.100.90.90 public fru-ctrl entity envmon

Thehost line points to the enabled traps.

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2-8Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

Enabling CallTracker

Caution The following is an example only. CallTracker must be used with care and in the propertroubleshooting scenarios, as it places considerable demands on the CPU. For a discussion ofCallTracker, refer to Managing Port Services on the Cisco AS5400 Universal Gateway at thefollowing URL:http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/nextport/dtnxptxd.htm

Step 1 Enable CallTracker

calltracker enablecalltracker history max-size 900calltracker history retain-mins 86400

Note Refer to Cisco IOS SNMP Traps Supported and How to Configure Them at the following URL:http://www.cisco.com/warp/public/477/SNMP/snmp_traps.html

Assigning RLM Groups

Step 1 Assign an RLM group and link weights. This determines where the RLM signaling comes from. SeeUsing Cisco RLM, page 3-17.

rlm group 0 protocol rlm port 3002 server columbia link address 10.40.0.10 source FastEthernet0/0 weight 2 link address 10.41.0.10 source FastEthernet0/1 weight 1 server fairfield link address 10.40.0.11 source FastEthernet0/0 weight 2 link address 10.41.0.11 source FastEthernet0/1 weight 1

Assigning Multiple RLM Groups

Voice gateways in release 2.0 of the Cisco SS7 Interconnect for Voice Gateways can support up to eightRLM groups per gateway. This capability enables you to spread trunks over multiple gateways. In orderfor the Cisco PGW 2200 to distinguish among the different RLM groups on each gateway, you mustassign a unique UDP port number to each RLM group on a gateway.

Step 1 Configure the RLM groups as appropriate for your network.

interface Serial1/0/0:23 ip unnumbered Loopback0 dialer pool-member 1 isdn switch-type primary-ni isdn incoming-voice modem isdn calling-number 333444333

isdn rlm-group 1

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no isdn send-status-enquiry isdn negotiate-bchan isdn bchan-number-order ascending

interface Serial1/0/3:23 ip unnumbered Loopback0 dialer pool-member 1 isdn switch-type primary-ni isdn incoming-voice modem isdn calling-number 333444333 isdn rlm-group 2 no isdn send-status-enquiry isdn negotiate-bchan isdn bchan-number-order ascending

Step 2 Assign link weights to the first RLM group.

rlm group 1 server fifi link address 10.4.8.10 source Loopback1 weight 90

Note The default UDP port for RLM groups is 3000. There is no need to change this for the firstRLM group.

Step 3 Assign link weights and a unique UDP port to each remaining RLM group.

rlm group 2 protocol rlm port 3001 server fifi link address 10.4.8.10 source Loopback2 weight 90

Note Be sure to include the corresponding UDP ports when you configure RLM groups on theCisco PGW 2200.

Assigning RADIUS Server Hosts and Ports

Step 1 Assign RADIUS server hosts and ports.

radius-server host 10.100.40.40 auth-port 1645 acct-port 1646 retransmit 3 key ciscoradius-server host 10.100.50.50 auth-port 1645 acct-port 1646 retransmit 3 key ciscoradius-server host 10.100.15.20 auth-port 1645 acct-port 1646 retransmit 3 key ciscoradius-server retransmit 3radius-server key cisco

Step 2 Configure the gateway to use VSAs.

radius-server vsa send accountingradius-server vsa send authentication

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2-10Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

Enabling Call Treatment and IVR

Step 1 Enable call treatment. SeeCall Admission Control and RSVP, page 3-2.

call treatment on <---enables call treatmentcall threshold global cpu-5sec low 50 high 75 <---sets thresholdscall rsvp-sync

Note The above does not use RAI (Resource Allocation Indicators). Not all Call Admission Control (CAC)features are available in early releases of the Cisco SS7 Interconnect for Voice Gateways 2.0.

Step 2 Do something similar to the following to enable interactive voice response (IVR) prompts for prepaidcalling-card services.

a. Declare the location of the Cisco TCL IVR scripts on a TFTP server.

call application voice debit tftp://10.100.10.10/tcl/app_debitcard.2.0.0.tcl

b. Determine a user ID length.

call application voice debit uid-len 4

c. In our example, English will be the first language of choice.

call application voice debit language 1 en

d. Spanish will be the second language.

call application voice debit language 2 sp

e. This is the location of the two prompt files, respectively.

call application voice debit set-location en 0 tftp://10.100.10.10/prompts/en/call application voice debit set-location sp 0 tftp://10.100.10.10/prompts/sp/

Assigning Dial Peers to Voice Ports

Step 1 Assign dial peers to voice ports.

Here is a POTS voice dial peer.

voice-port 1/0:1:D!

dial-peer voice 901 pots incoming called-number 902....... no shutdown destination-pattern 901110[0-4]... direct-inward-dial port 1/0:1:D prefix 901!

Here is a VoIP dial peer.

dial-peer voice 901103 voip destination-pattern 901103.... session target ras

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2-11Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

!

Here is a dial peer for prepaid calling-card services.

dial-peer voice 69 pots description voice Prepaid application debit incoming called-number 8006661234 destination-pattern 8006661234 port 1/0:1:D!

Here are more VoIP dial peers.

dial-peer voice 903 voip destination-pattern 903....... session target ras!dial-peer voice 9021092 voipdestination-pattern 9021092... session target ras!dial-peer voice 9021090 voipdestination-pattern 9021090... session target ras codec g711ulaw

Enabling RAI

Step 1 Globally set a CAC H.323 RAI resource threshold on all ports. This also causes RAI information to besent to the GK.

call threshold global cpu-avg low 90 high 95 busyoutgateway resource threshold high 90 low 85

Caution The above values should be appropriate for most situations. However, an issue related to ISDN causecodes must be taken into account. A cause code is sent once the high call threshold is crossed and thechannels are in the process of transitioning from an IS (in-service) busy or IS idle state to an OOS(out-of-service) state. Before the channels go into an OOS state (which can take seconds to occur),any TDM call that attempts to connect to these channels will be rejected with a cause code of 41(temporary failure).

Configuring a GatekeeperConfiguring the gatekeeper is straightforward.

Step 1 Establish a time zone.

clock timezone edt -4ip subnet-zero

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Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

Step 2 Enable the Cisco VoIP CAC with RSVP feature. SeeCall Admission Control and RSVP, page 3-2.

call rsvp-sync!interface FastEthernet0/0 ip address 10.41.7.50 255.255.0.0 duplex full!interface FastEthernet1/0 ip address 10.40.7.50 255.255.0.0 no ip mroute-cache duplex full ntp broadcast client!ip default-gateway 10.100.10.10ip classlessip route 0.0.0.0 0.0.0.0 10.40.0.1no ip http server

Step 3 Establish a gatekeeper identity, as well as zones, gateway priorities, and a technology prefix.

gatekeeper zone local z1-gk1 voice 10.40.7.50 zone remote z2-gk1 voice 10.70.7.50 1719 zone remote z3-gk1 voice 10.80.7.50 1719 zone prefix z1-gk1 901103* gw-priority 10 z1-gw1 zone prefix z1-gk1 901103* gw-priority 0 z1-gw2 z1-gw3 zone prefix z1-gk1 901108* gw-priority 10 z1-gw2 zone prefix z1-gk1 901108* gw-priority 0 z1-gw1 z1-gw3 zone prefix z1-gk1 901110* gw-priority 10 z1-gw3 zone prefix z1-gk1 901110* gw-priority 0 z1-gw1 z1-gw2 zone prefix z2-gk1 902* zone prefix z3-gk1 903* gw-type-prefix 1#* default-technology no shutdown

Configuring TDM Switching ServicesUpon receiving an incoming call with SS7, ISDN PRI, or CAS signaling, Cisco voice gateways analyzethe dialed digits and, if required, forward the call outward (using the appropriate outbound signaling) tothe designated port or trunk group. This feature, variously referred to as “grooming,” “drop and insert,”or (in EMEA) “tromboning,” is necessary for PSTN interconnects to provide not only legacy voiceservices but also test calls. The TDM switching feature of the gateways allow cross-connections to bemade directly on the time slot interchange (TSI) portion of the DSP.

Any Cisco AS5000 series trunk interface (T1 or E1, including T1s inside a CT3) can be designated asan outbound or inbound trunk for TDM switching purposes. SS7, network-side ISDN PRI, user-sideISDN PRI, or CAS signaling is provided on this outbound trunk to signal calls redirected by the gateway.Calls to be redirected are identified simply through a dial-peer match of the called number, or DNIS(Dialed Number Identification Service).

Refer also to the following useful documents at their respective URLs:

• Configuring Voice over IP

http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/multi_c/mcprt1/mcdvoip.htm

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• Network Side ISDN PRI Signalling, Trunking, and Switching

http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtpri_ni.htm

Example Configuration

The example configuration presented below illustrates one way to configure both SS7-to-PRI andPRI-to-SS7 switching. Controllers 1/0:1 through 1/0:14 represent SS7 ingress and egress facilities.Controllers 1/0:15 through 1/0:28 represent ISDN non-NFAS ingress and egress facilities, with a switchtype of NI2 (National ISDN-2).

In this example, any incoming SS7 10-digit call with the called-number NPA-NXX digits 904-102 isswitched to the ISDN PRIs on controllers 1.0:15 through 1/0:28. Conversely, any incoming 10-digit callon the ISDN PRIs with the called-number NPA-NXX digits 904-704 is switched to the SS7 RLM NFASgroup 0, which consists of controllers 1/0:1 through 1/0:14.

On the ISDN PRI egress side, this configuration provides two principal benefits:

• It uses dial-peer hunting to group a large number of T1 ISDN PRI spans into a single hunt group.

• It minimizes B-channel glare when TDM switched calls are processed in both directions, by usingB-channel negotiation and opposing B-channel hunt schemes.

Step 1 Assign the T3 controller.

controller T3 1/0 framing m23 clock source line cablelength 133 t1 1-28 controller description T3 to 5800_D1002

Step 2 Assign T1 controllers to serve SS7 NFAS ingress and egress facilities.

T1 1/0:1 framing esf pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 0!controller T1 1/0:2 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 1 nfas_group 0!<---snip---> T1 controllers 1/0:3 through 1/0:12 not shown

controller T1 1/0:13 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 12 nfas_group 0!controller T1 1/0:14 framing esf pri-group timeslots 1-24 nfas_d none nfas_int 13 nfas_group 0

Step 3 Assign T1 controllers to serve ISDN non-NFAS ingress and egress facilities.

controller T1 1/0:15 framing esf pri-group timeslots 1-24!controller T1 1/0:16 framing esf pri-group timeslots 1-24

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<---snip---> controllers 1/0:17 through 1/0:26 not shown

controller T1 1/0:27 framing esf pri-group timeslots 1-24!controller T1 1/0:28 framing esf pri-group timeslots 1-24

Step 4 Assign a serial controller to support the SS7 D-channel. This supports the RLM link for non-ISDNsignaling over IP.

interface Serial1/0:1:23ip unnumbered Loopback0encapsulation pppdialer pool-member 2no snmp trap link-statusisdn switch-type primary-niisdn incoming-voice modemisdn rlm-group 0no isdn send-status-enquiryisdn negotiate-bchan resend-setup <---Important! See Caution below

Caution Configure B-channel negotiation to support simultaneous ingress and egress traffic.

ppp authentication chap!

Step 5 Assign serial controllers to support the ISDN D-channels. This also automatically creates the voiceports.

interface Serial1/0:15:23ip unnumbered Loopback0encapsulation pppdialer pool-member 2no snmp trap link-statusisdn switch-type primary-niisdn incoming-voice modemisdn T306 30000isdn T310 4000isdn negotiate-bchan resend-setupisdn bchan-number-order ascending <--- Important! See Caution below

Caution To reduce the chance of B-channel glare (assignment contention) with bidirectional traffic, make thenear-end hunt proceed in a direction opposite that of the far-end setting. As the default isdescending,it is most likely the setting at the far end. However, take care to confirm the far-end hunt directionfirst.

no cdp enableppp authentication chap!interface Serial1/0:16:23no ip addressencapsulation pppdialer pool-member 2no snmp trap link-statusisdn switch-type primary-niisdn incoming-voice modemisdn T306 30000isdn T310 4000

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2-15Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

isdn negotiate-bchan resend-setupisdn bchan-number-order ascendingidle-character marksno cdp enableppp authentication chap!<---snip---> serial controllers 1/0:17 through 1/0:26 not shown

interface Serial1/0:27:23no ip addressencapsulation pppdialer pool-member 2no snmp trap link-statusisdn switch-type primary-niisdn incoming-voice modemisdn T306 30000isdn T310 4000isdn bchan-number-order ascendingisdn negotiate-bchan resend-setupidle-character marksno cdp enableppp authentication chap!interface Serial1/0:28:23no ip addressencapsulation pppdialer pool-member 2no snmp trap link-statusisdn switch-type primary-niisdn incoming-voice modemisdn T306 30000isdn T310 4000isdn bchan-number-order ascendingisdn negotiate-bchan resend-setupidle-character marksno cdp enableppp authentication chap

Note The following voice ports appear in the configuration, but they do not have to be assigned. The voiceports are created automatically when the ISDN serial D-channels are created.

!voice-port 1/0:1:D!voice-port 1/0:15:D!voice-port 1/0:16:D

<---snip---> voice-ports 1/0:17 through 1/0:26 not shown

voice-port 1/0:27:D!voice-port 1/0:28:D!dial-peer voice 1 pots description SS7 to PRI TDM switching <---see Note below

Note This dial peer begins the SS7-to-PRI hunt. By default its preference is 0, but there is nopreference 0line.

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Chapter 2 Voice Networks: Design FundamentalsGateway Configuration Examples

incoming called-number 904102.... <---periods represent wildcards destination-pattern 904102.... no digit-strip direct-inward-dial port 1/0:15:D forward-digits all!dial-peer voice 2 pots <--- second peer in the hunt, with preference 1 description SS7 to PRI TDM switching preference 1 incoming called-number 904102.... destination-pattern 904102.... no digit-strip direct-inward-dial port 1/0:27:D forward-digits all!dial-peer voice 3 pots <---third peer in the hunt, with preference 2 description SS7 to PRI TDM switching preference 2 incoming called-number 904102.... destination-pattern 904102.... no digit-strip direct-inward-dial port 1/0:16:D forward-digits all!<---snip---> dial-peers 4 through 9 not shown

dial-peer voice 10 pots <---tenth peer in the hunt, with preference 9 description SS7 to PRI TDM switching preference 9 incoming called-number 904102.... destination-pattern 904102.... no digit-strip direct-inward-dial port 1/0:23:D forward-digits all!dial-peer voice 11 pots <---eleventh peer in the hunt, with preference 10 description SS7 to PRI TDM switching preference 10 incoming called-number 904102.... destination-pattern 904102.... no digit-strip direct-inward-dial port 1/0:24:D forward-digits all!dial-peer voice 12 pots <---see Note below

Note This dial peer does not have a preference. It switches calls from the ISDN PRIs to the SS7 T1controllers 1/0:1 through 1/0:14, established inStep 2.

description PRI to SS7 TDM switching peer incoming called-number 904704.... destination-pattern 904704.... no digit-strip direct-inward-dial port 1/0:1:D

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2-17Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsManaging Echo Cancellation

forward-digits all

Managing Echo Cancellation

OverviewIn Cisco gateways, echo cancellation is enabled by default, with tail-delay coverage set at 8milliseconds. However, if you are using the Cisco echo canceller in the gateways, it is important todetermine the maximum echo-path tail delay and IP network delay that may exist in your network. Inaddition, other services (such as wireless) may add additional echo-path delays. If echo delay is longerthan the provisioned tail length, echo cancellation will not work.

In general, you should enable echo cancellation in networks where predicted echo-path delays exceed32 milliseconds. Also, if you plan to use external echo cancellation, Cisco recommends that you disablethe echo cancellers in the gateways. This will save memory and other platform resources.

Note Information about echo cancellation terminology and guidelines for network design can be found inITU recommendation G.168, available athttp://www.itu.org. See also Echo Analysis for Voice overIP at the following URL:http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/ea_isd.htm

The following example echo-cancellation configurations are presented:

• Disabling Echo Cancellation

• Changing Tail-Delay Coverage

• Typical Echo-Cancellation Settings

Disabling Echo Cancellation

Step 1 Issue the following commands to disable echo cancellation on a voice port:

Caution Because voice ports are created automatically when an ISDN D-channel or CAS signaling is assignedto a controller, you must determine which voice ports require echo cancellation and which do not. Inthis SS7 example there is only one voice port, as SS7 requires the instantiation of only one voice port(1/0:1:D), to support the serial D-channel.

5400#conf t5400(config-voiceport)#voice-port 1/0:1:D5400(config-voiceport)#no echo-cancel enable5400(config-voiceport)#

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Chapter 2 Voice Networks: Design FundamentalsManaging Echo Cancellation

Changing Tail-Delay CoverageIf you decide to use echo cancellation in your gateways, you may have different needs regarding thetail-delay coverage setting.

Step 1 Issue the following commands to change the tail-delay coverage setting:

5400#conf t5400(config)#voice-port 1/0:1:D5400(config-voiceport)#echo-cancel coverage5400(config-voiceport)#echo-cancel coverage ? 128 128 milliseconds echo canceller coverage 16 16 milliseconds echo canceller coverage 24 24 milliseconds echo canceller coverage 32 32 milliseconds echo canceller coverage 64 64 milliseconds echo canceller coverage 8 8 milliseconds echo canceller coverage

5400(config-voiceport)#echo-cancel coverage 128 <---See Note below.

Note This sets the echo canceller to cover a tail delay of 128 milliseconds.

5400(config-voiceport)#exit

Typical Echo-Cancellation SettingsHere are some typical echo-cancellation settings, most of which are defaults. In this SS7 case, thesettings are mapped from this port to a port related to a path in the echo canceller.

5400#sho voice port

ISDN 1/0:1:D - 1/0:1:D <--serial D-channel for SS7 RLM group (T1 controllers 1/0:1-1/0:14) Type of VoicePort is ISDN Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Non Linear Mute is disabled Non Linear Threshold is -21 dB Music On Hold Threshold is Set to -38 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancellation NLP mute is disabled Echo Cancellation NLP threshold is -21 dB Echo Cancel Coverage is set to 128 ms Playout-delay Mode is set to default Playout-delay Nominal is set to 60 ms Playout-delay Maximum is set to 200 ms Playout-delay Minimum mode is set to default, value 40 ms Playout-delay Fax is set to 300 ms Connection Mode is normal Connection Number is not set

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Chapter 2 Voice Networks: Design FundamentalsManaging Echo Cancellation

Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Call Disconnect Time Out is set to 60 s Ringing Time Out is set to 180 s Wait Release Time Out is set to 30 s Companding Type is u-law Region Tone is set for US Station name None, Station number None

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2-20Cisco SS7 Interconnect for Voice Gateways 2.0: Implementation Guide

Chapter 2 Voice Networks: Design FundamentalsManaging Echo Cancellation