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 63.122.238.180
ip name-server 65.211.120.197
ip name-server 63.122.233.180
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 0.0.0.0 208.254.78.190
!
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.2.0.0.0.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:esp_incoming_fax@blah.net
!
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 199.249.19.1
!
end

Thanks,
Mike.

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 ipv4:26.0.0.2
    !

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


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