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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:
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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. |
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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. |
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
Beginning with Release 2.0 of the Cisco SS7 Interconnect for Voice Gateways Solution, Cisco supports network 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 following conditions:
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Note These recommendations do not guarantee 100% call completion rates or uninterrupted service from SS7 links. To ensure the highest levels of service, Cisco continues to recommend that the Cisco SLT be colocated with the Cisco PGW 2200. |
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
The Generic Transparency Descriptor (GTD) for Gatekeeper Transaction Message Protocol (GKTMP) feature provides additional functionality to voice gateways and gatekeepers in a Cisco SS7 Interconnect for 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 messages of existing SS7 ISUP protocols in text format and allows SS7 messages to be carried as a payload in the H.225 registration, admission, and status (RAS) messages between the GW and GK. GTD messages can also be transported between GWs and GKs in H.323 messages. With the GTD feature, the GK extracts the GTD message and the external route server derives routing and accounting information based upon the 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, destination number, and port source. Adding support for SS7 ISUP messages allows the VoIP network to use additional 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/122 xu2/ftgtdpay.htm
Voice wholesalers use multiple ingress and egress carriers to route traffic. A call coming in to a gateway on a particular ingress carrier must be routed to an appropriate egress carrier. As networks grow and become more complicated, the dial plans needed to route the carrier traffic efficiently become more complex and the need for carrier sensitive routing (CSR) increases. The Gateway Trunk and Carrier Based Routing Enhancements feature implements CSR for Cisco voice gateways. Gateway feature enhancements add the following routing features:
For detailed instructions on configuring gateway trunk and carrier based routing enhancements, refer to documentation at the following URL:
The following configuration examples are presented, with commentary:
The 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 presentation of the following examples is that of their appearance in the configuration file. Not all steps are required in all networks.
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Note The Cisco SS7 Interconnect for Voice Gateways Solution is |
The following configuration examples are presented:
Management Information Bases, or MIBs, can be used to manage data for a variety of purposes.
Step 2 Set MIB parameter options. The following example applies to ISDN service.
call-history-mib retain-timer 60 call-history-mib max-size 500 !
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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 |
The following assigns T3 and T1 controllers and NFAS groups.
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Note Refer to Chapter 3, "Basic Configuration Using the Command-Line Interface," of Cisco AS5350 and
Cisco 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 100 t1 1-28 controller
Step 2 Assign T1 controllers and NFAS groups. Multiple spans (controllers) will use the same D-channel. The D-channel configuration is on interface serial 1/0:1:23. See also Configuring TDM Switching Services.
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Note Refer to Chapter 3, "Basic Configuration Using the Command-Line Interface," of Cisco AS5350 and
Cisco 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 for ISDN 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 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
gw-accounting h323 vsa <---required for prepaid service only gw-accounting voip
interface Loopback0 ip address 10.44.4.4 255.255.255.0
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Note Before performing this step, be sure to check for existing loopback interfaces with the show interface loopback command. |
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#
interface FastEthernet0/1 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
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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 10.100.20.20 tacacs-server host 10.100.30.30 tacacs-server host 10.101.30.30 tacacs-server timeout 3 <---recommended; see Caution below tacacs-server key cisco tacacs-server administration
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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 RW snmp-server enable traps snmp authentication linkdown linkup coldstart warmstart snmp-server enable traps fru-ctrl snmp-server enable traps entity snmp-server enable traps envmon snmp-server host 10.100.90.90 public fru-ctrl entity envmon
The host line points to the enabled traps.
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Caution The following is an example only. CallTracker must be used with care and in the proper troubleshooting scenarios, as it places considerable demands on the CPU. For a discussion of CallTracker, refer to Managing Port Services on the Cisco AS5400 Universal Gateway at the following URL: http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/nextport/dtnxptxd.ht m |
calltracker enable calltracker history max-size 900 calltracker history retain-mins 86400
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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 |
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
Voice gateways in release 2.0 of the Cisco SS7 Interconnect for Voice Gateways can support up to eight RLM groups per gateway. This capability enables you to spread trunks over multiple gateways. In order for the Cisco PGW 2200 to distinguish among the different RLM groups on each gateway, you must assign a unique UDP port number to each RLM group on a gateway.
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 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
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Note The default UDP port for RLM groups is 3000. There is no need to change this for the first RLM 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
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Note Be sure to include the corresponding UDP ports when you configure RLM groups on the Cisco PGW 2200. |
radius-server host 10.100.40.40 auth-port 1645 acct-port 1646 retransmit 3 key cisco radius-server host 10.100.50.50 auth-port 1645 acct-port 1646 retransmit 3 key cisco radius-server host 10.100.15.20 auth-port 1645 acct-port 1646 retransmit 3 key cisco radius-server retransmit 3 radius-server key cisco
Step 2 Configure the gateway to use VSAs.
radius-server vsa send accounting radius-server vsa send authentication
call treatment on <---enables call treatment call threshold global cpu-5sec low 50 high 75 <---sets thresholds call rsvp-sync
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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 prepaid calling-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/
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 !
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 voip destination-pattern 9021092... session target ras ! dial-peer voice 9021090 voip destination-pattern 9021090... session target ras codec g711ulaw
call threshold global cpu-avg low 90 high 95 busyout gateway resource threshold high 90 low 85
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Caution The above values should be appropriate for most situations. However, an issue related to ISDN cause codes must be taken into account. A cause code is sent once the high call threshold is crossed and the channels 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 the gatekeeper is straightforward.
clock timezone edt -4 ip subnet-zero
Step 2 Enable the Cisco VoIP CAC with RSVP feature. See Call Admission Control and RSVP.
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 10.100.10.10 ip classless ip route 0.0.0.0 0.0.0.0 10.40.0.1 no 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
Upon receiving an incoming call with SS7, ISDN PRI, or CAS signaling, Cisco voice gateways analyze the dialed digits and, if required, forward the call outward (using the appropriate outbound signaling) to the 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 voice services but also test calls. The TDM switching feature of the gateways allow cross-connections to be made 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 as an outbound or inbound trunk for TDM switching purposes. SS7, network-side ISDN PRI, user-side ISDN 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:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121cgcr/multi_c/mcprt1/mcdvoip.htm
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtpri_ni.htm
The example configuration presented below illustrates one way to configure both SS7-to-PRI and PRI-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 switch type of NI2 (National ISDN-2).
In this example, any incoming SS7 10-digit call with the called-number NPA-NXX digits 904-102 is switched to the ISDN PRIs on controllers 1.0:15 through 1/0:28. Conversely, any incoming 10-digit call on the ISDN PRIs with the called-number NPA-NXX digits 904-704 is switched to the SS7 RLM NFAS group 0, which consists of controllers 1/0:1 through 1/0:14.
On the ISDN PRI egress side, this configuration provides two principal benefits:
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 <---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-ISDN signaling over IP.
interface Serial1/0:1:23 ip unnumbered Loopback0 encapsulation ppp dialer pool-member 2 no snmp trap link-status isdn switch-type primary-ni isdn incoming-voice modem isdn rlm-group 0 no isdn send-status-enquiry isdn negotiate-bchan resend-setup <---Important! See Caution below
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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 voice ports.
interface Serial1/0:15:23 ip unnumbered Loopback0 encapsulation ppp dialer pool-member 2 no snmp trap link-status isdn switch-type primary-ni isdn incoming-voice modem isdn T306 30000 isdn T310 4000 isdn negotiate-bchan resend-setup isdn bchan-number-order ascending <--- Important! See Caution below
no cdp enable ppp authentication chap ! interface Serial1/0:16:23 no ip address encapsulation ppp dialer pool-member 2 no snmp trap link-status isdn switch-type primary-ni isdn incoming-voice modem isdn T306 30000 isdn T310 4000 isdn negotiate-bchan resend-setup isdn bchan-number-order ascending idle-character marks no cdp enable ppp authentication chap ! <---snip---> serial controllers 1/0:17 through 1/0:26 not shown interface Serial1/0:27:23 no ip address encapsulation ppp dialer pool-member 2 no snmp trap link-status isdn switch-type primary-ni isdn incoming-voice modem isdn T306 30000 isdn T310 4000 isdn bchan-number-order ascending isdn negotiate-bchan resend-setup idle-character marks no cdp enable ppp authentication chap ! interface Serial1/0:28:23 no ip address encapsulation ppp dialer pool-member 2 no snmp trap link-status isdn switch-type primary-ni isdn incoming-voice modem isdn T306 30000 isdn T310 4000 isdn bchan-number-order ascending isdn negotiate-bchan resend-setup idle-character marks no cdp enable ppp authentication chap
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Note The following voice ports appear in the configuration, but they do not have to be assigned. The voice ports 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
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Note This dial peer begins the SS7-to-PRI hunt. By default its preference is 0, but there is no preference 0 line. |
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
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Note This dial peer does not have a preference. It switches calls from the ISDN PRIs to the SS7 T1 controllers 1/0:1 through 1/0:14, established in Step 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 forward-digits all
In Cisco gateways, echo cancellation is enabled by default, with tail-delay coverage set at 8 milliseconds. However, if you are using the Cisco echo canceller in the gateways, it is important to determine the maximum echo-path tail delay and IP network delay that may exist in your network. In addition, other services (such as wireless) may add additional echo-path delays. If echo delay is longer than the provisioned tail length, echo cancellation will not work.
In general, you should enable echo cancellation in networks where predicted echo-path delays exceed 32 milliseconds. Also, if you plan to use external echo cancellation, Cisco recommends that you disable the echo cancellers in the gateways. This will save memory and other platform resources.
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Note Information about echo cancellation terminology and guidelines for network design can be found in
ITU recommendation G.168, available at http://www.itu.org. See also
Echo Analysis for Voice over IP 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:
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Caution Because voice ports are created automatically when an ISDN D-channel or CAS signaling is assigned to a controller, you must determine which voice ports require echo cancellation and which do not. In this 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 t 5400(config-voiceport)#voice-port 1/0:1:D 5400(config-voiceport)#no echo-cancel enable 5400(config-voiceport)#
If you decide to use echo cancellation in your gateways, you may have different needs regarding the tail-delay coverage setting.
5400#conf t 5400(config)#voice-port 1/0:1:D 5400(config-voiceport)#echo-cancel coverage 5400(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.
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Note This sets the echo canceller to cover a tail delay of 128 milliseconds. |
5400(config-voiceport)#exit
Here are some typical echo-cancellation settings, most of which are defaults. In this SS7 case, the settings 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 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
Posted: Tue Mar 12 16:27:55 PST 2002
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