Cisco 3745 modem PPP termination via VoIP
Category: Computers > Internet
Asked by: mrself-ga
List Price: $200.00
25 Nov 2006 06:07 PST
Expires: 25 Dec 2006 06:07 PST
Question ID: 785434
I presently use a Cisco 3745 as a "TDM to VoIP" gateway. Voice telephone calls made on 7 ISDN PRI's on the 3745 are converted to SIP signalling and G.711 RTP and terminated onto a carrier's network for final call completion. Here's a "graphical" representation of a current call flow: (PBX(ISDN PRI) -> Cisco3754(converted to SIP/VoIP) -> PSTN Network) The 3745 is merely acting as a TDM to SIP/VoIP signalling and media gateway. Here is the beginning of a "SHOW VERSION" for s/w revs: Cisco IOS Software, 3700 Software (C3745-ADVIPSERVICESK9-M), Version 12.3(4)YE1, RELEASE SOFTWARE (fc1) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2004 by Cisco Systems, Inc. Compiled Thu 30-Sep-04 15:44 by kellmill ROM: System Bootstrap, Version 12.2(8r)T2, RELEASE SOFTWARE (fc1) Here is my TWO PART question (both parts must be answered) #1 - Without interrupting the presently functionality, is the 3745 capable of taking incoming MODEM calls (i.e. modems calling in to connect PPP) via SIP/VoIP? IOW, I would like to take calls that are coming from MODEMS and terminate them into the 3745 via SIP/VoIP just like a regular access server/modem bank would answer, negotiate/handshake, assign an IP address and PPP session and allow the modem caller to access our network. (In yet another way of asking this, just for CLARITY, I would like to use the 3745 as a remote access server but connecting to modems via SIP/VoIP rather than the traditional ISDN, T1 or local phone line method). Hope this is clear! #2 - What changes need to be made to the following configuration to enable this? (Note that all identifying information has been changed to protect our network security). Current configuration : 11735 bytes ! ! No configuration change since last restart ! version 12.3 service timestamps debug datetime msec localtime show-timezone service timestamps log datetime localtime show-timezone service password-encryption ! hostname HOSTNAME ! boot-start-marker boot-end-marker ! logging buffered 4096 notifications enable secret 5 blahblahblah enable password 7 blahblahblah ! clock timezone GMT 0 voice-card 1 dspfarm dsp services dspfarm ! voice-card 2 dspfarm ! voice-card 3 dspfarm ! voice-card 4 dspfarm ! no aaa new-model ip subnet-zero ! ! ! ! ip domain name blah.NET ip name-server 18.104.22.168 ip name-server 22.214.171.124 ip name-server 126.96.36.199 ip ssh break-string ip audit notify log ip audit po max-events 100 no ftp-server write-enable isdn switch-type primary-dms100 ! ! voice rtp send-recv ! voice service pots ! voice service voip fax protocol pass-through g711ulaw sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 rel1xx disable ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /\(^\+\)1\(..........\)/ /\0/ rule 2 /^1\(..........\)/ /+1\1/ rule 3 /........../ /+1\0/ rule 4 /......./ /+1555\0/ rule 5 /..../ /+1555555\0/ rule 6 /.../ /+15555555\0/ rule 7 /../ /+155555555\0/ rule 8 /./ /+1555555555\0/ rule 9 /^$/ /+15555555555/ ! ! voice translation-profile rpid translate calling 1 ! ! voip-incoming translation-profile rpid ! ! ! ! crypto isakmp policy 1 hash md5 authentication pre-share crypto isakmp key NUMEROUS CRYPTO TUNNEL ENTRIES ! crypto ipsec security-association lifetime seconds 86400 ! crypto ipsec transform-set ipcom ah-md5-hmac ! crypto map sip local-address FastEthernet0/0 crypto map sip 1 ipsec-isakmp description RTO set peer xx.xx.xx.xx set transform-set ipcom set pfs group2 match address 120 crypto map sip 2 ipsec-isakmp description ELB set peer xx.xx.xx.xx set transform-set ipcom set pfs group2 match address 121 ! ! ! controller T1 1/0 framing esf crc-threshold 320 clock source line primary linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! controller T1 1/1 framing esf crc-threshold 320 linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! controller T1 2/0 framing esf crc-threshold 320 linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! controller T1 2/1 framing esf crc-threshold 320 linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! controller T1 3/0 framing esf crc-threshold 320 linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! controller T1 3/1 framing esf crc-threshold 320 linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! controller T1 4/0 framing esf crc-threshold 320 linecode b8zs cablelength short 133 pri-group timeslots 1-24 ! ! ! interface FastEthernet0/0 ip address xx.xx.xx.xx 255.255.255.192 speed auto full-duplex no mop enabled crypto map sip ! interface FastEthernet0/1 ip address xx.xx.xx.xx 255.255.255.192 shutdown duplex auto speed auto ! interface Serial1/0:23 no ip address no logging event link-status isdn switch-type primary-dms100 isdn timer t321 30000 isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! interface Serial1/1:23 no ip address no logging event link-status isdn switch-type primary-dms100 isdn timer t321 30000 isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! interface Serial2/0:23 no ip address no logging event link-status isdn switch-type primary-dms100 isdn timer t321 30000 isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! interface Serial2/1:23 no ip address no logging event link-status isdn switch-type primary-dms100 isdn timer t321 30000 isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! interface Serial3/0:23 no ip address no logging event link-status isdn switch-type primary-dms100 isdn timer t321 30000 isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! interface Serial3/1:23 no ip address no logging event link-status isdn switch-type primary-dms100 isdn timer t321 30000 isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! interface Serial4/0:23 no ip address no logging event link-status isdn switch-type primary-dms100 isdn timer t321 30000 isdn incoming-voice voice isdn guard-timer 1000 isdn send-alerting no fair-queue no cdp enable ! ip classless ip route 0.0.0.0 0.0.0.0 188.8.131.52 ! no ip http server no ip http secure-server ! ! access-list 120 permit ip host xx.xx.xx.xx yy.yy.yy.yy 0.0.0.31 snmp-server community public RO snmp-server enable traps tty ! ! ! ! control-plane ! ! call application voice ONRAMP flash:app_faxmail_onramp.184.108.40.206.tcl ! ! voice-port 1/0:23 translation-profile incoming rpid no echo-cancel enable ! voice-port 1/1:23 translation-profile incoming rpid no echo-cancel enable ! voice-port 2/0:23 translation-profile incoming rpid no echo-cancel enable no comfort-noise ! voice-port 2/1:23 translation-profile incoming rpid ! voice-port 3/0:23 translation-profile incoming rpid no echo-cancel enable ! voice-port 3/1:23 translation-profile incoming rpid no echo-cancel enable ! voice-port 4/0:23 translation-profile incoming rpid no echo-cancel enable ! ! ! ! ! dial-peer voice 10 pots application session direct-inward-dial port 1/0:23 ! dial-peer voice 100 voip tone ringback alert-no-PI description voip dial peer - eveything goes to the NS preference 5 application session destination-pattern .T rtp payload-type cisco-codec-fax-ack 114 rtp payload-type cisco-codec-fax-ind 113 rtp payload-type nte 98 voice-class codec 1 session protocol sipv2 session target sip-server fax-relay ecm disable fax rate 14400 ip qos dscp cs5 media ip qos dscp cs3 signaling no vad ! dial-peer voice 11 pots application session direct-inward-dial port 1/1:23 ! dial-peer voice 20 pots application session direct-inward-dial port 2/0:23 ! dial-peer voice 21 pots application session direct-inward-dial port 2/1:23 ! dial-peer voice 30 pots application session direct-inward-dial port 3/0:23 ! dial-peer voice 31 pots application session direct-inward-dial port 3/1:23 ! dial-peer voice 40 pots application session direct-inward-dial port 4/0:23 ! dial-peer voice 1 voip voice-class codec 1 ip udp checksum ! dial-peer voice 99 mmoip application fax_on_vfc_onramp_app out-bound max-conn 200 destination-pattern 0T information-type fax session target mailto:firstname.lastname@example.org ! dial-peer voice 98 pots application onramp incoming called-number 0T direct-inward-dial forward-digits 0 ! dial-peer voice 999 voip preference 4 application session destination-pattern 9999999999 rtp payload-type cisco-codec-fax-ack 114 rtp payload-type cisco-codec-fax-ind 113 rtp payload-type nte 98 voice-class codec 1 session protocol sipv2 session target sip-server fax-relay ecm disable fax rate 14400 ip qos dscp cs5 media ip qos dscp cs3 signaling no vad ! sip-ua no remote-party-id set sip-status 400 pstn-cause 31 set sip-status 401 pstn-cause 21 set sip-status 403 pstn-cause 21 set sip-status 405 pstn-cause 63 set sip-status 406 pstn-cause 79 set sip-status 410 pstn-cause 22 set sip-status 488 pstn-cause 31 set sip-status 501 pstn-cause 38 set sip-status 503 pstn-cause 41 set sip-status 606 pstn-cause 38 set pstn-cause 6 sip-status 406 set pstn-cause 27 sip-status 502 set pstn-cause 30 sip-status 501 set pstn-cause 31 sip-status 480 set pstn-cause 43 sip-status 502 set pstn-cause 44 sip-status 503 set pstn-cause 49 sip-status 503 set pstn-cause 50 sip-status 503 set pstn-cause 58 sip-status 503 set pstn-cause 63 sip-status 503 set pstn-cause 66 sip-status 480 set pstn-cause 69 sip-status 503 set pstn-cause 70 sip-status 503 set pstn-cause 81 sip-status 502 set pstn-cause 82 sip-status 502 set pstn-cause 83 sip-status 503 set pstn-cause 84 sip-status 503 set pstn-cause 85 sip-status 503 set pstn-cause 86 sip-status 408 set pstn-cause 88 sip-status 503 set pstn-cause 91 sip-status 502 set pstn-cause 95 sip-status 503 set pstn-cause 96 sip-status 409 set pstn-cause 97 sip-status 480 set pstn-cause 98 sip-status 409 set pstn-cause 99 sip-status 480 set pstn-cause 100 sip-status 501 set pstn-cause 101 sip-status 503 set pstn-cause 111 sip-status 500 retry invite 2 retry bye 2 retry cancel 2 sip-server dns:blah.blah.com ! ! alias exec u undebug all alias exec sr show run ! line con 0 line aux 0 line vty 0 4 password 7 57943756984375983275893 logout-warning 3600 login ! warm-reboot ntp clock-period 17177117 ntp server 220.127.116.11 ! end Thanks, Mike.
Re: Cisco 3745 modem PPP termination via VoIP
Answered By: leapinglizard-ga on 25 Dec 2006 05:14 PST
Dear mrself, You may indeed be able to transfer modem calls over VoIP. This is generally not possible because the voice compression used in most VoIP systems irretrievably degrades the modem signal. However, with a non-compressed protocol such as G.711 on a high-quality network, the signal may be preserved and used by a suitable installation. You have indicated that you do use G.711, and your high bandwidth capacity (7 PRIs) suggests that you may have the requisite quality as well. The problem is that the codecs used by VOIP ATAs are designed to compress voice, not the analog signals sent and received by modems. A second problem is if a non-compressing codec is used, the transmission will be very sensitive to network QoS, i.e., packet loss, jitter, and latency will be issues. To successfully use data modems over a VOIP connection you will need a minimum of: * A non-compressing codec - ITU G.711 is the usual choice * A very high-quality network connection voip-info.org: Modem over VOIP http://www.voip-info.org/wiki/view/Modem+over+VOIP G.711 pass-through transports a modem signal as audio encoded samples across a network at a constant rate of 8,000 eight-bit samples per second. This equates to a constant rate of 64,000 bit/s. If reliability against IP impairments is included by means of redundancy, IP bandwidth requirements can double or even triple when using the pass-through method. CommsDesign: MoIP: Making PSTN Modems Work on IP Networks http://www.commsdesign.com/design_corner/OEG20030312S0017 Cisco documentation indicates that modem passthrough is available on your 3745 router with the addition of a High-Density Digital Voice/Fax Network Module. The network modules provide enterprises, managed service providers, and service providers with the ability to directly connect the public switched telephone network (PSTN), traditional telephony equipment (such as private branch exchange (PBX), key systems, analog telephones, and fax machines), and WAN to Cisco 2600XM, Cisco 2691, 2811, 2821, 2851, Cisco 3700 Series and Cisco 3800 Series Access Routers for either IP Communications capabilities or pure toll bypass. [...] Important Features and Benefits [...] Fax and modem passthrough: Allows fax and modem traffic to pass through a voice port. Cisco: Cisco 3700 Series Multiservice Access Routers: IP Communications Voice/Fax Network Module Datasheet http://www.cisco.com/en/US/products/hw/routers/ps282/products_data_sheet09186a0080191d41.html Your IOS version is 12.3, and modem passthrough has been supported at least since the 12.1 release. The Modem Passthrough over VoIP feature performs the following functions: * Represses processing functions like compression, echo cancellation, high-pass filter, and voice activity detection (VAD). * Issues redundant packets to protect against random packet drops. * Provides static jitter buffers of 200 milliseconds to protect against clock skew. * Discriminates modem signals from voice and fax signals, indicating the detection of the modem signal across the connection, and placing the connection in a state that transports the signal across the network with the least amount of distortion. * Reliably maintains a modem connection across the packet network for a long duration under normal network conditions. Cisco: Modem Passthrough over Voice over IP: Feature Overview http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtmodptr.htm#1015329 With modem passthrough, the phone signal is converted into packets and transferred to its destination over the IP network. It must then be converted back into a phone signal for reception by a modem. When the gateway detects a data modem, both the originating gateway and the terminating gateway roll over to G.711. The roll over to G.711 disables the high-pass filter, disables echo cancellation, and disables VAD. At the end of the modem call, the voice ports revert to the prior configuration and the digital signal processor (DSP) goes back to the state before switchover. Cisco: Modem Passthrough over Voice over IP: Passthrough Switchover http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtmodptr.htm#1022993 You need to configure modem passthrough in both the originating gateway and the terminating gateway for the Modem Passthrough over VoIP feature to operate. If you configure only one of the gateways in a pair, the modem call will not connect successfully. Cisco: Modem Passthrough over Voice over IP: Configuration Tasks http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtmodptr.htm#63234 The exact configuration will depend on your setup. The following should give you an idea of the necessary changes. voice-port 0:D ! dial-peer voice 1 pots incoming called-number 55511.. destination-pattern 020.. direct-inward-dial port 0:D prefix 020 ! dial-peer voice 2 voip incoming called-number 020.. destination-pattern 55511.. modem passthrough nse codec g711ulaw redundancy session target ipv4:18.104.22.168 ! Cisco: Modem Passthrough over Voice over IP: Configuration Examples http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtmodptr.htm#1026322 An alternative to using modem passthrough is to add an Analog Modem Network Module to your 3745 router. In this way, you incorporate a modem into the 3745, so that modem calls are converted from the PSTN into a PPP session at the router. By combining a T1 channel service unit/data service unit (CSU/DSU) WAN interface card, an analog modem network module, and possibly voice over IP, the Cisco 2600, 3600 and 3700 can provide a one-chassis solution to all branch office requirements. [...] Cisco IOS dial access software-Cisco IOS software provides a broad range of features for the dial-in user, including: ?- Autosensing IPX, TCP/IP, AppleTalk Remote Access (ARA), AppleTalk Control Protocol (ATCP) ?- Serial Line Internet Protocol (SLIP), PPP, and MP ?- Reverse Telnet support for LAN-based dial-out ?- Domain Name System (DNS) Domain Name Server support These features enable a wide variety of dial-in clients to utilize the applications and facilities of the branch office network. Cisco: Cisco 2600/3600/3700 Series Analog Modem Network Modules: Data Sheet http://www.cisco.com/en/US/products/hw/routers/ps259/products_data_sheet09186a00801b1c38.html Only modem passthrough lets you take advantage of free global IP to carry your modem calls, but then you will need a gateway at each end of the connection. Adding the Analog Modem Network Module lets your 3745 accept modem calls from any client on the PSTN. Regards, leapinglizard Search strategy: cisco ios 12.3 modem pass-through ://www.google.com/search?q=cisco+ios+12.3+modem+pass-through cisco modem pass-through ://www.google.com/search?q=cisco+modem+pass-through modem over voip cisco ://www.google.com/search?q=modem+voip+cisco
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