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Table Of Contents
Cisco V.150.1 Minimum Essential Requirements (For Cisco IOS Release 15.1(4)M Only)
Prerequisites for Cisco V.150.1 MER
Restrictions for Cisco V.150.1 MER
Information About Cisco V.150.1 MER
Differences Between Cisco V.150.1 MER and Cisco Legacy V.150.1
Advantages of Modem Relay Over Modem Pass-through
SCIP—EI, Modem over IP, and Fax over IP Interfaces
Cisco V.150.1 MER Network Architecture
How to Configure Cisco V.150.1 MER
Configuring the Gateway in the Cisco Unified CM Administration
Configuring the Phone Settings in the Cisco Unified CM Administration
Adding a New Directory Number in the Cisco Unified CM Administration
Configuring the Gateway (Line-side)
Configuring SCCP on Cisco IOS Gateways
Configuring Modem Transport Methods for STCAPP Devices
Configuring Modem Pass-through Calls
Configuring V.150.1 Modem Relay Parameters
Configuring the Gateway (Trunk-side)
Configuring the T1 Controller and Operating Parameters
Configuring MGCP for Compatibility with Cisco UCM
Configuring MGCP Parameters for Modem Relay
Configuring the Profile-level V.150.1 Filter
Associating a SIP Trunk Security Profile with a Trunk
Setting the Service Parameter-level V.150.1 Filter
Verifying and Troubleshooting the Cisco V.150.1 MER Configuration
Symptoms and Possible Solutions for Cisco V.150.1 MER
Feature Information for Cisco V.150.1 Minimum Essential Requirements
Cisco V.150.1 Minimum Essential Requirements (For Cisco IOS Release 15.1(4)M Only)
First Published: March 25, 2011Last Updated: March 25, 2011
Note
The information in this document applies to the Cisco V.150.1 Minimum Essential Requirements feature beginning in Cisco IOS Release 15.1(4)M (dated March 28, 2011). This feature was originally released as the Secure Communication Between IP-STE Endpoint and Line-Side STE Endpoint feature in Cisco IOS Releases 12.4(4)T and 12.4(9)T. For reference purposes, there is some "legacy" information provided here about the original feature. For more detailed information about the original feature, see the Secure Communication Between IP-STE Endpoint and Line-Side STE Endpoint document.
The Cisco V.150.1 Minimum Essential Requirements feature complies with the requirements of the National Security Agency (NSA) SCIP-216 Minimum Essential Requirements (MER) for V.150.1 recommendation. The SCIP-216 recommendation has simplified the existing V.150.1 requirements. Beginning in Cisco IOS Release 15.1(4)M, the Cisco V.150.1 MER feature adds negotiation support to the following interfaces:
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This feature also provides support for Unified Capability Requirement (UCR) 2008 Modem over IP (MoIP) and Fax over IP (FoIP).
The V.150.1 is an ITU recommendation for using a modem over IP networks that support dialup modem calls for large installed bases of modems and telephony devices operating on a traditional public switched telephone network (PSTN). The V.150.1 recommendation specifically defines how to relay data from modems and telephony devices on a PSTN into and out of an IP network via a modem.
In Cisco IOS Release 12.4(4)T, Cisco developed the Secure Communication Between IP Secure Endpoint and Trunk-Side Secure Terminal Equipment (STE) Endpoint feature, and in Cisco IOS Release 12.4(9)T, the Secure Communication Between IP Secure Endpoint and Line-Side STE Endpoint feature to meet the requirements of this standard. In this document, these features are referred to as "Cisco Legacy V.150.1."
This document focuses primarily on the capabilities of the Cisco V.150.1 MER feature in Cisco IOS Release 15.1(4)M, but also provides some information for the Cisco Legacy V.150.1 feature.
Contents
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Prerequisites for Cisco V.150.1 MER
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Restrictions for Cisco V.150.1 MER
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Information About Cisco V.150.1 MER
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Cisco Legacy V.150.1
In Cisco IOS Release 12.4(4)T, the Secure Communication Between IP Secure Endpoint and Line-Side STE Endpoint feature enabled V.150.1 for STCAPP-control voice ports and allowed an on-network secure terminal equipment (STE), connected directly to a Cisco IOS gateway, to establish a secure call to an IP secure endpoint. Figure 1 shows a basic topology for the V.150.1 standard.
Figure 1
Standard Topology for the V.150.1 Standard
In Cisco IOS Release 12.4(9)T, the Secure Communication Between IP Secure Endpoint and Trunk-Side STE Endpoint feature implemented V.150.1 for the Cisco IOS gateway. The capability was implemented only on MGCP gateways for placing secure calls between the IP secure endpoints and off-network STE devices via MGCP-controlled time-division multiplexing (TDM) trunks.
STE utilizes both modem pass-through and modem relay for secure phone calls. Cisco and another company implemented V.150.1 to carry SCIP (formerly known as Future Narrow Band Digital Terminal [FNBDT]) data to meet the DoD requirements of STE. There is also a VoIP STE that uses only modem relay for secure phone calls.
Cisco's Legacy V.150.1 implementation contains the following features:
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Differences Between Cisco V.150.1 MER and Cisco Legacy V.150.1
Table 1 summarizes the differences and advantages of Cisco V.150.1 MER over the Cisco Legacy V.150.1.
Note
Table 2 summarizes the hardware and software compatibility information for Cisco Legacy V.150.1 and Cisco V.150.1 MER.
Advantages of Modem Relay Over Modem Pass-through
The advantages of modem relay over modem pass-through are:
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For more information, see Fax/Modem over IP.
SCIP—EI, Modem over IP, and Fax over IP Interfaces
The following interfaces are supported for SCIP and MoIP:
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The following interfaces are supported for FoIP:
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The following interface is not supported for the UCR 2008 SCIP, MoIP, FoIP functionality, provided by this feature:
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Cisco V.150.1 MER Network Architecture
The two types of endpoints in the MER network are:
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The Cisco V.150.1 MER network architecture (shown in Figure 2) supports the following:
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Figure 2
Cisco V.150.1 MER Network Architecture
How to Configure Cisco V.150.1 MER
To configure the line-side functionality of the Cisco V.150.1 MER feature, perform the following tasks:
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Configuring the Cisco UCM
To configure the Cisco UCM, perform the tasks in this section.
Step 1
Step 2
Step 3
Step 4
Step 5
Step 6
Step 7
Step 8
Configuring the Gateway in the Cisco Unified CM Administration
To configure the gateway, perform the tasks in this section. See Figure 3 for an example screen of gateway configuration settings.
Step 1
Step 2
Step 3
Step 4
Step 5
Step 6
Step 7
Step 8
Figure 3
Example of Gateway Configuration Settings
Configuring the Phone Settings in the Cisco Unified CM Administration
To configure the phone settings, perform the following steps. Figure 4 provides an example of the screen for phone configuration.
Step 1
Note
Step 2
Step 3
Figure 4
Example of Phone Configuration Settings
Adding a New Directory Number in the Cisco Unified CM Administration
To add a new directory number (DN), perform the task in this section. Figure 5 and Figure 6 provide examples of the screens for DN settings.
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Step 2
Step 3
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Step 5
Figure 5
Example of Directory Number (DN) Settings (Part 1)
Figure 6
Example of Directory Number Settings (Part 2)
Configuring the Gateway (Line-side)
To configure the line-side gateway, perform the following tasks (in some of these tasks, the command syntax has been abbreviated for clarity):
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Configuring SCCP on Cisco IOS Gateways
SCCP messaging enables Cisco Unified Communications Manager endpoint call control using the STCAPP. To configure SCCP on the Cisco IOS gateway, perform the tasks in this section.
SUMMARY STEPS
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DETAILED STEPS
Step 1
enable
Example:Router> enable
Enables privileged EXEC mode.
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Step 2
configure terminal
Example:Router# configure terminal
Enters global configuration mode.
Step 3
sccp local interface-type interface-number
Example:Router(config)# sccp local fastethernet 0/0
Selects the local interface that the SCCP application uses to register with Cisco Unified Communications Manager.
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Step 4
sccp ccm {ip-address | dns} identifier identifier-number [port port-number] [version version-number]
Example:Router(config)# sccp ccm 10.1.1.1 version 8
Adds a Cisco UCM server to the list of available servers and sets various parameters.
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sccp
Example:Router(config)# sccp
Enables SCCP and its related applications.
Step 6
sccp ccm group group-number
Example:Router(config)# sccp ccm group 1
Creates a Cisco UCM group.
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Step 7
associate ccm identifier-number priority priority-number
Example:Router(config)# associate ccm 1 priority 1
Associates a Cisco UCM with a Cisco UCM group.
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Step 8
exit
Example:Router(config)# exit
Exits the current configuration mode.
Configuring Modem Transport Methods for STCAPP Devices
This task configures modem transport methods for STCAPP devices. Perform this task to specify modem transport capability.
SUMMARY STEPS
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DETAILED STEPS
Configuring Modem Pass-through Calls
This task configures modem pass-through calls on the gateway. Perform this task to enable interoperation with the SCCP gateway running versions of Cisco IOS software prior to Cisco IOS Release 12.4(4)T that are not V.150.1-capable.
SUMMARY STEPS
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DETAILED STEPS
Configuring V.150.1 Modem Relay Parameters
This task configures optional V.150.1 modem-relay parameters. Configure these parameters to address specific network conditions for latency, redundancy, and V.14 parameters.
SUMMARY STEPS
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DETAILED STEPS
Configuring the Gateway (Trunk-side)
To configure the trunk side of the gateway, perform the following tasks:
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Configuring the T1 Controller and Operating Parameters
To configure the T1 controller and operating parameters, perform the tasks in this section.
SUMMARY STEPS
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DETAILED STEPSConfiguring MGCP for Compatibility with Cisco UCM
To ensure proper operation of the Cisco V.150.1 MER feature on the Cisco UCM, peform the MGCP CLI configuration steps in this section.
SUMMARY STEPS
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DETAILED STEPSConfiguring MGCP Parameters for Modem Relay
To configure MGCP parameters for modem relay, perform the tasks in this section.
SUMMARY STEPS
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DETAILED STEPS
Configuring the SIP Trunk
SIP SDP content includes information from both Legacy Cisco V.150 and V.150.1 MER, and SIP options include the Profile-level V.150.1 Filter and Service Parameter-level V.150.1 Filter. For a chart showing modem transport methods, see Table 3 in the "Troubleshooting Tips" section. To configure the SIP trunk, perform the following tasks:
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Configuring the Profile-level V.150.1 Filter
To configure the profile-level V.150.1 filter, perform the tasks in this section. Figure 7 provides a sample screen of this configuration procedure.
Step 1
Step 2
Step 3
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Step 5
Step 6
Step 7
Step 8
Step 9
Step 10
Step 11
Associating a SIP Trunk Security Profile with a Trunk
To associate a SIP trunk security profile with a trunk, complete the tasks in this section. Figure 7 provides a sample screen of the SIP trunk profile configuration.
Step 1
Step 2
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Figure 7
Example of SIP Trunk Security Profile Configuration
Setting the Service Parameter-level V.150.1 Filter
To set the service parameter-level V.150.1 filter, perform the tasks in this section.
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Troubleshooting Tips
The following options are provided to fix interoperability issues that may arise due to some additions made to the SDP content to ensure backward compatibility with existing Cisco UCMs running Cisco Legacy V.150.1. Although according to the V.150.1 specification these additions should not impact SDP parsing, the fail-safe option to remove them is provided. These options can be configured on a per-trunk or per-cluster basis:
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Table 3 Chart of Modem Transport Methods
voice band data 1
voice band data
voice band data
None
voice band data
voice band data
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V.150.1 modem relay
voice band data
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V.150.1 Modem Relay
V.150.1 modem relay
voice band data
voice band data
voice band data
V.150.1 modem relay
None
V.150.1 modem relay
V.150.1 modem relay
IP
1 voice band data (VDB) = modem Pass-through
What to Do Next
For more information on configuring SIP trunks in Cisco Unified Communications Manager 8.0(2), see Understanding Cisco Unified Communications Manager Trunk Types.
For additional information about SIP and configuring SIP trunks, see Understanding Session Initiation Protocol.
Verifying and Troubleshooting the Cisco V.150.1 MER Configuration
To verify and troubleshoot the configuration of the Cisco V.150.1 MER feature, perform the steps in this section. The show commands provide information about the configuration. The debug commands are useful when problems are apparent in the system. The information in Step 10 provides guidelines for ensuring a correct configuration. Table 4 in Step 9 provides a list of symptoms that may occur and possible resolutions to those problems.
SUMMARY STEPS
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DETAILED STEPS
Step 1
Use the show voice dsp active command to display status information for all DSP voice channels:
Router# show voice dsp active----------------------------FLEX VOICE CARD 1 -----------------------*DSP ACTIVE VOICE CHANNELS*DSP DSPWARE VOX DSP SIG DSP PAK TX/RXTYPE VERSION CODEC NUM CH TS VOICEPORT SLT NUM CH TS RST AI ABRT PACK COUNT====== ========== ======== === == == ========= === === == == === == ==== ===C5510 28.0.136 modem-rel 001 01 05 1/0/0 001 001 04 06 0 0 0 2912/3533C5510 28.0.136 modem-rel 001 02 23 1/0:23 001 002 11 23 0 0 0 3458/3021------------------------END OF FLEX VOICE CARD 1 --------------------Step 2
Use the show call active voice command to display call information for voice calls in progress:
Router# show call active voice...Modem Relay Mode = signaling-assistedModem Relay Local Rx Speed=9600 bpsModem Relay Local Tx Speed=9600 bpsModem Relay Remote Rx Speed=19200 bpsModem Relay Remote Tx Speed=19200 bpsModem Relay Phy Layer Protocol=v32Modem Relay Ec Layer Protocol=v14SPRTInfoFramesReceived=0SPRTInfoTFramesSent=0SPRTInfoTFramesResent=0SPRTXidFramesReceived=0SPRTXidFramesSent=1SPRTTotalInfoBytesReceived=806778SPRTTotalInfoBytesSent=806562SPRTPacketDrops=0Step 3
Use the show stcapp device voice-port 1/0/0 command to display call information for voice calls on a specific port:
Router# show stcapp device voice-port 1/0/0Port Identifier: 1/0/0Device Type: ALGDevice Id: 6Device Name: AN1A6D001760200Device Security Mode : NoneModem Capability: BothDevice State: ISDiagnostic: NoneDirectory Number: 2011Dial Peer(s): 100Dialtone after remote onhook feature: activatedBusytone after remote onhook feature: not activatedLast Event: STCAPP_CC_EV_CALL_FEATURELine State: ACTIVELine Mode: CALL_BASICHook State: OFFHOOKmwi: DISABLEvmwi: OFFmwi config: BothPrivacy: Not configuredPLAR: DISABLECallback State: DISABLEDCWT Repetition Interval: 0 second(s) (no repetition)Number of CCBs: 1Global call info:Total CCB count = 1Total call leg count = 2Call State for Connection 1 (ACTIVE): TsConnectedConnected Call Info:Call Reference: 28055870Call ID (DSP): 52Local IP Addr: 10.10.10.139Local IP Port: 18258Remote IP Addr: 10.10.10.139Remote IP Port: 17748Calling Number: 2011Called Number: 3011Codec: g711ulawSRTP: offMER Capabilites Active:Capability and Version : 0x20110000Modulation and RFC2833 : 0xF0000005SPRT Max Payload Chan0 : 0SPRT Max Payload Chan2 : 0SPRT Max Payload Chan3 : 0SPRT Max WinSize Chan2 : 0SSE Standard Support : 0x5SSE Vendor Support : 0x5NSE Payload Value : 0RFC2833 Payload Value : 101SSE Payload Value : 0SPRT Payload Value : 0NoAudio Payload Value : 0Step 4
Use the debug voice application stcapp all command to display debugging information for the components of the STCAPP:
Router# debug voice application stcapp all*Jan 4 20:45:50.877: 1/0/0: Registering device*Jan 4 20:45:50.877: 1/0/0: stcapp_register_device...*Jan 4 20:45:51.881: sccp_parse_control_msg: glob_ccm->version 9*Jan 4 20:45:51.881: SCCP(AN43E17E8B90200)rcvd RegisterAckMessage*Jan 4 20:45:51.881: sccp_appl_service_stop_timer: Stop A69DA3C timer*Jan 4 20:45:51.881: sccp_parse_control_msg_v1: rcvd register ack, ka_interval 30, for prof_id 0, appl_type 4 negotiated sccp version 21*Jan 4 20:45:51.881: RegisterAck msg rcvd in hex -81 0 0 0 1E 0 0 0 4D 2F 44 2F 59 0 0 0 3C 0 0 0 15 20 F1 FF...Jan 4 20:45:51.881: sccp_parse_control_msg: glob_ccm->version 9*Jan 4 20:45:51.881: SCCP(AN43E17E8B90200)rcvd CapabilitiesReqMessage*Jan 4 20:45:51.881: sccp_generate_msg: msg_id 16 msg_len 296 pak_size 304*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: Codec list with pkt_period (cnt 16) -*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 257*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 257257 30,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 112*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 112112 20,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 114*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 114114 220,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 299*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 299299 20,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 300*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 300*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: v150_mr.cap_n_ver: 0x1120*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: v150_mr.mod_n_2833: 0xFF0F00F0300 0,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 301*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 301*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: v150_sprt_payload.chan0_max_payload: 35840*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: v150_sprt_payload.chan2_max_payload: 33792*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: v150_sprt_payload.chan3_max_payload: 35840*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: v150_sprt_payload.chan2_max_windows: 2048301 0,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 302*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 302*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: v150_sse.standdard_field 0x5000000302 0,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 111*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 111111 20,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 113*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 113113 220,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 4*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 44 20,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 2*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 22 20,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 11*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 1111 220,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 12*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 1212 220,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 15*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 1515 220,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 11*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 1111 220,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_rec->codec = 86*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: msg_cap->payload_caps = 86*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: codec_params=0300000086 120,*Jan 4 20:45:51.881: sccp_send_capabilities_rsp_msg_v21: CapRes msg txed in hex(including header) - pak->datagramsize 304, actual_len 272*Jan 4 20:45:51.881: sccp_print_hex_msg: Len:272 Hex:28 01 00 00 15 00 00 00 10 00 00 00 10 00 00 00 01 01 00 00 1E 00 00 00 00 00 00 00 00 00 00 00 70 00 00 00 14 00 00 00 00 00 00 00 00 00 00 00 72 00 00 00 DC 00 00 00 00 00 00 00 00 00 00 00 2B 01 00 00 14 00 00 00 00 00 00 00 00 00 00 00 2C 01 00 00 00 00 00 00 00 00 11 20 FF 0F 00 F0 2D 01 00 00 00 00 00 00 8C 00 84 00 8C 00 08 00 2E 01 00 00 00 00 00 00 05 00 00 00 00 00 00 00 6F 00 00 00 14 00 00 00 00 00 00 00 00 00 00 00 71 00 00 00 DC 00 00 00 00 00 00 00 00 00 00 00 04 00 00 00 14 00 00 00 00 00 00 00 00 00 00 00 02 00 00 00 14 00 00 00 00 00 00 00 00 00 00 00 0B 00 00 00 DC 00 00 00 00 00 00 00 00 00 00 00 0C 00 00 00 DC 00 00 00 00 00 00 00 00 00 00 00 0F 00 00 00 DC 00 00 00 00 00 00 00 00 00 00 00 0B 00 00 00 DC 00 00 00 00 00 00 00 00 00 00 00 56 00 00 00 78 00 00 00 03 00 00 00 00 00 00 00Step 5
The debug voice application stcapp all can also be used to display debug information for call setup on the line-side:
Router# debug voice application stcapp all...*Jan 4 20:56:33.266: sccp_parse_control_msg: glob_ccm->version 9*Jan 4 20:56:33.266: SCCP(AN43E17E8B90200)rcvd OpenReceiveChannel*Jan 4 20:56:33.266: OpenReceviceChannel msg rcvd in hex -5 1 0 0 32 19 AC 1 35 0 0 1 14 0 0 0 4 0 0 0 0 0 0 0 0 0 0 0 32 19 AC 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 65 0 0 0 A 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 A A A 8B 0 0 0 0 0 0 0 0 0 0 0 0 A0 F 0 0 0 0 0 0 0 0 0 0 1 0 0 0 0 0 11 20 FF F 0 F0 0 0 0 0 0 0 0 0 5 0 0 0 0 0 0 0 0 65 0 0 0 0 0 0*Jan 4 20:56:33.266: OpenReceiveChannelMsg Info:conference_id = 28055858, pass_through_party_id = 16777269msec_pkt_size = 20, compression_type = 4qualifier_in.ecvalue = 0, g723_bitrate = 0, call_ref = 28055858stream_pass_through_id = 0, rfc2833_payload_type = 101codec_dynamic_payload = 0, codec_mode = 0Encryption Info :: algorithm_id 0, key_len 0, salt_len 0requestedAddrType = 0, source_ip_addr.ipAddrType = 0, source_ip_addr = 10.10.10.139, source_port_number = 4000,audio_level_adjustment = 0*Jan 4 20:56:33.266: v150 latent caps active:modem relay cap and version: 0x20110000 modulation and rfc2833: 0xF0000FFFsprt max payload for chan0: 0 chan2: 0 chan3: 0, max window for chan2: 0sse standard support filed: 0x5 vendor support filed: 0x0payload nse 0 rfc2833 101 sse 0 v150_sprt 0 noaudio 0*Jan 4 20:56:33.266: sccp_dcapi_extract_and_validate_srtp_context*Jan 4 20:56:33.266: STCAPP:stcapp_get_dcb_and_lcb*Jan 4 20:56:33.266: 1/0/0: stcapp_get_dcb_and_lcb*Jan 4 20:56:33.266: 1/0/0: stcapp_screen_api_event*Jan 4 20:56:33.266: 1/0/0: event:STCAPP_DC_EV_MEDIA_OPEN_RCV_CHNL received.*Jan 4 20:56:33.266: 1/0/0: stcapp_screen_open_rcv_chnl*Jan 4 20:56:33.266: 1/0/0: active_ccb=0x11544A0, media_state is NO_MEDIA*Jan 4 20:56:33.266: 1/0/0: ==> Received event:STCAPP_DC_EV_MEDIA_OPEN_RCV_CHNL*Jan 4 20:56:33.266: 1/0/0: Call State:PROCEEDING*Jan 4 20:56:33.266: 1/0/0: stcapp_open_rcv_chnl_eh*Jan 4 20:56:33.266: 1/0/0: call_ref=28055858*Jan 4 20:56:33.266: 1/0/0: stcapp_get_ccb_ptr*Jan 4 20:56:33.266: 1/0/0: received ORC: rcv payload=101*Jan 4 20:56:33.266: 1/0/0: stcapp_set_up_voip_leg*Jan 4 20:56:33.266: 1/0/0: stcapp_get_ccb_ptr*Jan 4 20:56:33.266: 1/0/0: In stcapp_set_up_voip_leg, local port allocated 21240*Jan 4 20:56:33.266: 1/0/0: stcapp_set_up_modem_parms*Jan 4 20:56:33.266: STCAPP:Codec: 5 ptime :20, codecbytes: 160*Jan 4 20:56:33.266: 1/0/0: CCM directive -> enabling MER modem relay*Jan 4 20:56:33.266: 1/0/0: MR parms: sprt_retries=12, sprt_latency=200, sprt_rx_v14_pb_hold_time=50, sprt_tx_v14_hold_time=20, sprt_tx_v14_hold_count=16, gw_xid=1, dictsize=1024, stringlen=32, compressdir=3, sse_red_interval=20, sse_red_pkt_count=3, sse_t1=1000, sse_retries=3, rfc2833_bitmap=0*Jan 4 20:56:33.266: 1/0/0: Info provided to RTPSPI - sess_mode:2, desired_qos 0, codec 5, pkt_period 20,*Jan 4 20:56:33.266: 1/0/0: rem_port 4000, lr_port 21240, dtmf_mode 400, rcv_nte 101 nte 0*Jan 4 20:56:33.266: 1/0/0: Sending ccIFCallSetupRequest for voip leg*Jan 4 20:56:33.266: 1/0/0: ccIFCallSetRequest returned voip call id:12*Jan 4 20:56:33.266: 1/0/0: MER modem relay configuration passed down ? call id:12 MR proto = 4*Jan 4 20:56:33.266: STCAPP:stcapp_find_ccb_by_call_id:ERROR:Invalid Call ID*Jan 4 20:56:33.266: 1/0/0: stcapp_conn_db_insert_ccb*Jan 4 20:56:33.266: 1/0/0: ccb=0x11544A0*Jan 4 20:56:33.266: 1/0/0: call ccCallConnect for voice call_id 11*Jan 4 20:56:33.266: 1/0/0: Media state is set to RECV_ONLY*Jan 4 20:56:33.266: 1/0/0: Sending dcDeviceOpenReceiveChannelAck*Jan 4 20:56:33.266: 1/0/0: ORChnlAck Info: codec:5, loc_ipaddr: 10.10.10.143, loc_port:21240, chnl_id:16777269*Jan 4 20:56:33.266: sccp_spi_orc_ack: enqueue spi evt SCCP_SPI_MEDIA_ORC_ACK, reg_name=AN43E17E8B90200*Jan 4 20:56:33.266: 1/0/0: New State = CONNECTING*Jan 4 20:56:33.270: STCAPP:Receive CC event:: call_id=12, ccb=0x11544A0*Jan 4 20:56:33.270: 1/0/0: ==> Received event:STCAPP_CC_EV_CALL_CONNECTED for CallId: 12*Jan 4 20:56:33.270: 1/0/0: Call State:CONNECTING*Jan 4 20:56:33.270: 1/0/0: stcapp_call_connected_eh*Jan 4 20:56:33.270: 1/0/0: stcapp_create_conference*Jan 4 20:56:33.270: 1/0/0: Sending ccConferenceCreate to Symphony*Jan 4 20:56:33.270: 1/0/0: Conference created. voice call id:11, voip call id:12*Jan 4 20:56:33.270: 1/0/0: No state change*Jan 4 20:56:33.270: sym_xapp_process_ccapi_events: minor is ZERO - should be non-zero for CCAPI event*Jan 4 20:56:33.270: sccp_generate_msg: msg_id 34 msg_len 40 pak_size 48*Jan 4 20:56:33.270: sccp_open_receive_channel_ack_v14: going to send ack to CCM - status 0, ipaddr 10.10.10.143, port 21240, conn_id 16777269, prof_id 0*Jan 4 20:56:33.270: sccp_open_receive_channel_ack_v14: OpenRecvChnlAck msg txed in hex(including header) - len 48*Jan 4 20:56:33.270: sccp_print_hex_msg: Len:48 Hex:28 00 00 00 15 00 00 00 22 00 00 00 00 00 00 00 00 00 00 00 0A 0A 0A 8F F1 1D CE 99 F2 7F E0 98 50 10 0A F4 F8 52 00 00 35 00 00 01 32 19 AC 01...*Jan 4 20:56:33.270: sccp_transmit_msg: sending on socket 5*Jan 4 20:56:33.274: sccp_parse_control_msg: msg_ptr 16127364, msg_len 172, msg_id 138*Jan 4 20:56:33.274: sccp_parse_control_msg: glob_ccm->version 9*Jan 4 20:56:33.274: SCCP(AN43E17E8B90200)rcvd StartMediaTransmission*Jan 4 20:56:33.274: StartMediaTrans msg rcvd in hex -8A 0 0 0 32 19 AC 1 35 0 0 1 0 0 0 0 A A A 8B 0 0 0 0 0 0 0 0 0 0 0 0 A6 41 0 0 14 0 0 0 4 0 0 0 B8 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 32 19 AC 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 65 0 0 0 A 0 0 0 0 0 0 0 0 0 0 0 1 0 0 0 0 0 11 20 FF F 0 F0 0 0 0 0 0 0 0 0 5 0 0 0 0 0 0 0 0 65 0 0 0 0 0 0*Jan 4 20:56:33.274: StartMediaTransmissionMsg Info:conference_id = 28055858, pass_through_party_id = 16777269msec_pkt_size = 20, compression_type = 4remote_ip_addr = 10.10.10.139, remote_port = 16806qualifier_out.precedence_value = 184, qualifier_out.ssvalue = 0qualifier_out.max_frames_per_pkt = 0, g723_bitrate = 0, call_ref = 28055858, stream_pass_through_id = 0 rfc2833_payload_type = 101codec_dynamic_payload = 0, codec_mode = 0Encryption Info :: algorithm_id 0, key_len 0salt_len 0*Jan 4 20:56:33.274: v150 latent caps active:modem relay cap and version: 0x20110000 modulation and rfc2833: 0xF0000FFFsprt max payload for chan0: 0 chan2: 0 chan3: 0, max window for chan2: 0sse standard support filed: 0x5 vendor support filed: 0x0payload nse 0 rfc2833 101 sse 0 v150_sprt 0 noaudio 0Step 6
Use the debug voip rtp session named command to display debug information for session establishment:
Router# debug voip rtp session named*Jan 4 21:04:25.675: s=DSP d=VoIP payload 0x65 ssrc 0x1F2A sequence 0x811B timestamp 0x21C875DF*Jan 4 21:04:25.675: Pt:101 Evt:34 Pkt:0B 00 00 <Snd>>>...*Jan 4 21:04:25.923: Pt:101 Evt:35 Pkt:0B 07 D0 <Snd>>>...*Jan 4 21:04:29.283: <<<Rcv> Pt:118 Evt:12 Pkt:01 D8 2CStep 7
Use the debug mgcp packets command to display debug registration information for MGCP trunks:
Router# debug mgcp packets*Jan 4 17:50:50.547 EDT: MGCP Packet received from 10.10.10.132:2427--->AUEP 1581 S1/DS1-0/5@MER-CCM2GW8.cisco.com MGCP 0.1F: X, A, I<---*Jan 4 17:50:50.547 EDT: MGCP Packet sent to 10.10.10.132:2427--->200 1581I:X: 0L: p:10-20, a:PCMU;PCMA;G.nX64;NoAudio;telephone-event, fmtp:"telephone-event 0-15", b:64, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, X+mdste/md:V150;V150merrelay, v:T;G;D;L;H;R;ATM;SST;FXR;PRE;X+mdste;FML: p:10-220, a:G.729;G.729a;G.729b;telephone-event, fmtp:"telephone-event 0-15", b:8, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, X+mdste/md:V150;V150merrelay, v:T;G;D;L;H;R;ATM;SST;FXR;PRE;X+mdste;FML: p:10-110, a:G.726-16;G.728;telephone-event, fmtp:"telephone-event 0-15", b:16, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, X+mdste/md:V150;V150merrelay, v:T;G;D;L;H;R;ATM;SST;FXR;PRE;X+mdste;FML: p:10-70, a:G.726-24;telephone-event, fmtp:"telephone-event 0-15", b:24, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, X+mdste/md:V150;V150merrelay, v:T;G;D;L;H;R;ATM;SST;FXR;PRE;X+mdste;FML: p:10-50, a:G.726-32;telephone-event, fmtp:"telephone-event 0-15", b:32, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, X+mdste/md:V150;V150merrelay, v:T;G;D;L;H;R;ATM;SST;FXR;PRE;X+mdste;FML: p:30-270, a:G.723.1-H;G.723;G.723.1a-H;telephone-event, fmtp:"telephone-event 0-15", b:6, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, X+mdste/md:V150;V150merrelay, v:T;G;D;L;H;R;ATM;SST;FXR;PRE;X+mdste;FML: p:30-330, a:G.723.1-L;G.723.1a-L;telephone-event, fmtp:"telephone-event 0-15", b:5, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL, X+mdste/md:V150;V150merrelay, v:T;G;D;L;H;R;ATM;SST;FXR;PRE;X+mdste;FMM: sendonly, recvonly, sendrecv, inactive, loopback, conttest, data, netwloop, netwtest<---Step 8
Use the debug mgcp packets command to display debugging information about call setup on the MGCP trunk:
Router# debug mgcp packetsa=cpar: a=T38FaxMaxDatagram:320a=cpar: a=T38FaxUdpEC:t38UDPRedundancy<---*Jan 4 17:43:32.611 EDT: MGCP Packet received from 10.10.10.132:2427--->CRCX 1573 S1/DS1-0/23@MER-CCM2GW8.cisco.com MGCP 0.1C: D000000001ac193b000000F500000003X: 17L: p:20, a:PCMU;telephone-event, fmtp:"telephone-event 0-15,32-35", s:off, t:b8, X+mdste/md:v150merrelayM: recvonlyR: D/[0-9ABCD*#]Q: process,loop<---*Jan 4 17:43:32.619 EDT: MGCP Packet sent to 10.10.10.132:2427--->200 1573 OKI: 4v=0c=IN IP4 10.10.10.139m=audio 18938 RTP/AVP 0 101 100 118a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15,32-35a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194,200-202a=rtpmap:118 v150fw/8000a=fmtp:118 1,3-4a=X-sqn:0a=X-cap: 1 audio RTP/AVP 100a=X-cpar: a=rtpmap:100 X-NSE/8000a=X-cpar: a=fmtp:100 192-194,200-202a=X-cap: 2 image udptl t38a=sqn:0a=cdsc: 1 audio RTP/AVP 0 101 100 118a=cdsc: 5 audio udpsprt 120a=cpar: a=sprtmap:120 v150mr/8000a=cpar: a=fmtp:120 mr=1;mg=0;CDSCselect=1;jmdelay=no;Versn=1.1;mrmods=1,3a=cdsc: 6 image udptl t38a=cpar: a=T38FaxVersion:3a=cpar: a=T38MaxBitRate:33600a=cpar: a=T38FaxRateManagement:transferredTCFa=cpar: a=T38FaxMaxBuffer:200a=cpar: a=T38FaxMaxDatagram:320a=cpar: a=T38FaxUdpEC:t38UDPRedundancy<---*Jan 4 17:43:32.659 EDT: MGCP Packet received from 10.10.10.132:2427--->MDCX 1574 S1/DS1-0/23@MER-CCM2GW8.cisco.com MGCP 0.1C: D000000001ac193b000000F500000003I: 4X: 17L: p:20, a:PCMU;telephone-event, fmtp:"telephone-event 32-35", s:off, t:b8, X+mdste/md:v150merrelayM: sendrecvS:v=0o=- 4 0 IN EPN S1/DS1-0/23@MER-CCM2GW8.cisco.coms=Cisco SDP 0t=0 0m=audio 17712 RTP/AVP 0 101 118c=IN IP4 10.10.10.139a=rtpmap:101 telephone-eventa=fmtp:101 32-35a=rtpmap:118 v150fw/8000a=fmtp:118 1,3a=sqn:0a=cdsc: 1 audio RTP/AVP 0 101 118a=cdsc: 4 audio udpsprt 120a=cpar: a=sprtmap:120 v150mr/8000a=cpar: a=fmtp:120 mr=1;mg=0;CDSCselect=1;jmdelay=no;Versn=1.1;mrmods=1,3<---*Jan 4 17:43:32.663 EDT: MGCP Packet sent to 10.10.10.132:2427--->200 1574 OK<---*Jan 4 17:43:38.579 EDT: MGCP Packet sent to 10.10.10.132:2427--->NTFY 714848268 *@MER-CCM2GW8.cisco.com MGCP 0.1X: 0O:<---*Jan 4 17:43:38.579 EDT: MGCP Packet received from 10.10.10.132:2427--->200 714848268<---Step 9
Use the debug mgcp all command to display session information for debugging the MGCP trunk:
Router# debug mgcp all*Jan 4 17:54:46.499 EDT: //53/0776534D8005/MGCP|S1/DS1-0/23|-1|-1/<VOICE>/mgcp_xlate_call_feature_type(1062):[lvl=2 ]mgcp_xlate_call_feature_type: feature 47*Jan 4 17:54:46.499 EDT: //-1/xxxxxxxxxxxx/MGCP/mgcp_cr_and_init_evt_node(4596):[lvl=1]$$$ the node pointer 71E1B348*Jan 4 17:54:46.499 EDT: //-1/xxxxxxxxxxxx/MGCP/mgcp_insert_node_to_preprocess_q(4518):[lvl=1]$$$enq to preprocess, qhead=71E1B348, qtail=71E1B348, count 1, evtptr=71E1B348*Jan 4 17:54:46.499 EDT: //53/0776534D8005/MGCP|S1/DS1-0/23|-1|-1/<VOICE>/xlate_ccapi_ev(600):[lvl=1]MGCP APP gets CC_EV_CALL_FEATURE event: major code=EV_MEDIA_EVT, minor_code(d)=121, minor_code=v150merrelay, *pkg=67108864...*Jan 4 17:54:54.963 EDT: //53/0776534D8005/MGCP|S1/DS1-0/23|-1|-1/<VOICE>/mgcp_remove_old_ack(714):[lvl=1]Removing ack: (trans ID 1600) : 200 1600 OKI: 5v=0c=IN IP4 10.10.10.139m=audio 17748 RTP/AVP 0 101 100 118a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15,32-35a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194,200-202a=rtpmap:118 v150fw/8000a=fmtp:118 1,3-4a=X-sqn:0a=X-cap: 1 audio RTP/AVP 100a=X-cpar: a=rtpmap:100 X-NSE/8000a=X-cpar: a=fmtp:100 192-194,200-202a=X-cap: 2 image udptl t38a=sqn:0a=cdsc: 1 audio RTP/AVP 0 101 100 118a=cdsc: 5 audio udpsprt 120a=cpar: a=sprtmap:120 v150mr/8000a=cpar: a=fmtp:120 mr=1;mg=**MSG 00002 TRUNCATED****MSG 00002 CONTINUATION #01**0;CDSCselect=1;jmdelay=no;Versn=1.1;mrmods=1,3a=cdsc: 6 image udptl t38a=cpar: a=T38FaxVersion:3a=cpar: a=T38MaxBitRate:33600a=cpar: a=T38FaxRateM0anagement:transferredTCFa=cpar: a=T38FaxMaxBuffer:200a=cpar: a=T38FaxMaxDatagram:320a=cpar: a=T38FaxUdpEC:t38UDPRedundancy*Jan 4 17:54:55.047 EDT: //53/0776534D8005/MGCP|S1/DS1-0/23|-1|-1/<VOICE>/mgcp_remove_old_ack(714):[lvl=1]Removing ack: (trans ID 1601) : 200 1601 OKStep 10
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Symptoms and Possible Solutions for Cisco V.150.1 MER
This section provides information about some possible problems or issues that m ay arise when you are configuring and operating the Cisco V.150.1 MER feature. Review the information in Table 4 for symptoms and possible solutions to help ensure operability of the Cisco V.150.1 MER feature in your network.
Note
Additional References
Related Documents
Cisco IOS commands
Voice commands
Information related to MGCP
Detailed information about implementing fax/modem over IP
Information about the Cisco Unified Communications Manager
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Information about MGCP
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Information about SCCP
Skinny Client Control Protocol
Information about using Cisco UCM on SIP trunks
Understanding Cisco Unified Communications Manager Trunk Types.
Standards
ITU-T V.150.1
SCIP-216
Minimum Essential Requirements (MER) for V.150.1 Gateways Publication, Revision 2.0, 2 November 2007
DoD UCR 2008
Changes to UCR 2008, Change 1, Section 5.3.2, Assured Services Requirements
MIBs
RFCs
RFC 2833
RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
Technical Assistance
Feature Information for Cisco V.150.1 Minimum Essential Requirements
Table 5 lists the release history for this feature.
Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Note
Glossary
ANS—ANSwering tone.
ANSam—ANSwering tone with amplitude modulation.
AS-SIP—Assured Services SIP.
BRI—Basic Rate Interface.
CAS—channel-associated signaling. The transmission of signaling information within the voice channel.
CCM—Cisco CallManager. For updated terminology, see Cisco UCM.
CLI—command-line interface.
CM—Communications Manager.
Cisco UCM—Cisco Unified Communications Manager.
codec—compressor/decompressor.
DoD—Department of Defense.
DN—directory number.
DNS—Domain Name System.
DSP—digital signal processor.
EI—end instrument.
FNBDT—Future Narrow Band Digital Terminal. This protocol is used for transmitting secure calls over V.32 and V.34 datapumps.
FoIP—Fax over IP.
FXS—Foreign Exchange Station.
g.711 and g.729—ITU standards for coding analog signals into digital signals, and for audio (speech) compression and decompression.
GW—Gateway (analog endpoints). This includes analog phones, analog secure phones, analog fax machines, and analog modems.
ICT—inter-cluster trunk.
IETF—Internet Engineering Task Force.
IP—Internet Protocol.
IP-STE—Internet Protocol—Secure Terminal Equipment. Specialized encryption-capable IP phones that communicate only over V.150.1 modem relay.
ISDN—Integrated Services Digital Network. A communication protocol offered by telephone companies that permits telephone networks to carry data, voice, and other source traffic.
ISR—Integrated Services Router. Cisco 28xx series and 38xx series router.
ISR G2—Integrated Services Router Generation 2. Cisco 29xx and 39xx series routers.
ITU—International Telecommunications Union.
LSC—Local Switch Controller.
MER—Minimal Essential Requirement. This is also referred to as NSA specification SCIP-216.
MGCP—Media Gateway Control Protocol. A control and signal protocol for converting audio signals carried on public switched telephone network (PSTN) circuits to data packets carried over the internet or other packet networks. See also Media Gateway Control Protocol Voiceband Data Package and General Purpose Media Descriptor Parameter Package from IETF.
Modem Relay Preferred Endpoint—MER-compatible endpoint that transitions to modem relay with transmitting voice information in the audio state. Example: data-only endpoint that does not support audio capabilities and transitions to modem relay.
MoIP—Modem over IP, also referred to V.150.1 Modem Relay.
NM—network module.
NoAudio—Mechanism for avoiding audio transmission during the audio state. A modem relay preferred endpoint can use NoAudio to identify that it does not support audio capabilities. See section 4.9 in Minimum Essential Requirements (MER) for V.150.1 Gateways Publication, Revision 2.
NSA—National Security Agency.
Passthrough—This term is also referred to as voice band data.
PRI—Primary Rate Interface. An ISDN interface to primary rate access. Primary rate access is a single 64-kbps D channel plus 23 (T1) or 30 (E1) B channels for voice or data.
PSTN—public switched telephone network. A worldwide network based on copper wires, fiber-optic cables, microwave transmissions, cellular networks, communications satellites, and undersea telephone cables connected by switching centers. PSTN originally carried analog voice data, and carries analog and digital data.
PVDM—Packet Voice DSP Module (also referred to as DSP).
PVDM2—Packet Voice DSP Module version 2 (used in ISRs, ISR G2s and PVDM3).
RIC—Reason Identifier Code.
RTP—Real-time Transport Protocol. This protocol is for transmitting real-time data such as audio and video.
SCCP—Skinny Client Control Protocol. Network terminal control messaging protocol between a skinny client and the Cisco Unified Communications Manager.
SCIP-216—Secure Communications Interoperability Protocol (NSA Specification SCIP-216).
SCIP-EI—Secure Communications Interoperability Protocol-End Instrument. This refers to any MER-compliant IP endpoint that conforms to SCIP-215 section 5.3.2.21.3.
SDP—Session Description Protocol. This is the format used to describe streaming media initialization parameters.
SIP—Session Initiation Protocol.
SPRT—Simple Packet Relay Transport.
SRTP—Secure Real-time Transport Protocol.
SSE—State Signaling Event.
STCAPP—SCCP Telephony Control Application.
STE—secure terminal equipment. This refers to specialized encryption-capable BRI/analog phones that can communicate over V.150.1 modem relay or over modem pass-through.
STU—secure terminal unit. This refers to specialized encryption-capable analog phones that operate only over NSE-based modem pass-through connections.
T.38—ITU recommendation for allowing transmission of fax in real time over IP networks.
TDM—time-division multiplexing (see also PSTN).
UCR—Unified Capability Requirement.
V.90—ITU standard for 56-Kbps modems.
V.92—ITU standard providing convenience and performance improvements for dialup modems including faster connect times, faster upload speeds, and V.44 data compression.
VBD—Voice Band Data (also referred to as modem pass-through).
VIC—voice interface card.
VoIP—Voice over IP. Enables a router to carry voice traffic, for example, telephone calls and faxes, over an IP network.
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Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.
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