Table Of Contents
Configuring Voice Functionality
Analog and Digital Voice Port Assignments
Session Initiation Protocol (SIP)
Media Gateway Control Protocol (MGCP)
Dual Tone Multi Frequency Relay
SCCP-Controlled Analog Ports with Supplementary Features
Unified Survival Remote Site Telephony
Verification of Voice Configuration
Configuring Voice Functionality
This chapter provides information about configuring voice functionality on the Cisco 880 Series Integrated Services Routers (ISRs). The following ISRs have voice gateway capability:
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IAD881B and IAD888B: 2 BRI ports
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This chapter contains the following topics:
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Voice Ports
Analog voice ports (Foreign Exchange Station (FXS) ports) connect routers in packet-based networks to 2-wire or 4-wire analog circuits in telephony networks. Two-wire circuits connect to analog telephone or fax devices, and four-wire circuits connect to PBXs.
Digital voice ports are ISDN basic rate interface (BRI) ports.
Analog and Digital Voice Port Assignments
Analog and digital voice port assignments vary by model number. Table 8-1 lists the Cisco 880 series ISRs and their voice port assignments.
Voice Port Configuration
To configure analog and digital voice ports, see the following documents:
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Call Control Protocols
The Cisco 880 Series ISR voice gateway models support the following call control protocols:
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Session Initiation Protocol (SIP)
Session Initiation Protocol (SIP) is a peer-to-peer, multimedia signaling protocol developed in the IETF (IETF RFC 2543). Session Initiation Protocol is ASCII-based. It resembles HTTP, and it reuses existing IP protocols (such as DNS and SDP) to provide media setup and teardown. See the Cisco IOS SIP Configuration Guide for more information.
For router configuration information under SIP, see the Basic SIP Configuration chapter of the Cisco IOS SIP Configuration Guide.
Cisco 880 Series ISR voice gateways provide voice security through SIP enhancements within the Cisco IOS Firewall. SIP inspect functionality (SIP packet inspection and detection of pin-hole openings) is provided, as well as protocol conformance and application security. The user is given more granular control on the policies and security checks applied to SIP traffic, and capability to filter out unwanted messages. For more information, see "Cisco IOS Firewall: SIP Enhancements: ALG and AIC" at http://www.cisco.com/en/US/docs/ios/security/configuration/guide/sec_sip_alg_aic.html.
Media Gateway Control Protocol (MGCP)
Media Gateway Control Protocol (MGCP) RFC 2705 defines a centralized architecture for creating multimedia applications, including Voice over IP (VoIP). See the Cisco IOS MGCP and Related Protocols Configuration Guide for more information.
Cisco 880 series voice gateway ISRs are configured primarily as residential gateways (RGWs) under MGCP. For residential gateway configuration information, see the Configuring an RGW section of the Basic MGCP Configuration chapter of the Cisco IOS MGCP and Related Protocols Configuration Guide.
H.323
International Telecommunications Union Recommendation H.323 defines a distributed architecture for creating multimedia applications, including Voice over IP. See the Cisco IOS H.323 Configuration Guide for more information about H.323.
For router configuration information, see the Configuring H.323 Gateways chapter of the Cisco IOS H.323 Configuration Guide.
Dial Peer Configuration
Configuring dial peers is the key to implementing dial plans and providing voice services over an IP packet network. Dial peers are used to identify call source and destination endpoints and to define the characteristics applied to each call leg in the call connection. For router configuration information, see Dial Peer Configuration on Voice Gateway Routers.
Other Voice Features
The Cisco 880 series voice gateway ISRs support the following voice features:
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Real-Time Transport Protocols
Real-Time Transport Protocol (RTP) provides end-to-end network transport functions for applications that transmit real-time data.
Cisco Real-Time Transport Protocol (cRTP) uses the RTP protocol to forward Cisco-proprietary payload types.
Secure Real-Time Transport Protocol (SRTP) defines an RTP profile providing encryption, authentication, and replay protection.
RTP is used primarily with DTMF relay and is configured under dial peer configuration. For information on configuring RTP payload types, see the Dual-Tone Multifrequency Relay section of Dial Peer Configuration on Voice Gateway Routers.
For information on configuring SRTP on SIP-controlled platforms, see the Configuring SIP Support for RTP chapter of the Cisco IOS SIP Configuration Guide.
For configuring RTP on MGCP-controlled platforms, see the Configuring an RGW section of the Basic MGCP Configuration chapter of the Cisco IOS MGCP and Related Protocols Configuration Guide.
Dual Tone Multi Frequency Relay
Using Dial Tone Multi Frequency (DTMF) Relay the local VoIP gateway listens for DTMF digits and sends the digits uncompressed as either RTP packets or H.245 packets to the remote VoIP gateway. The remote VoIP gateway regenerates the DTMF digits. This methodology prevents digit loss due to compression. For information on configuring DTMF Relay, see the Dual-Tone Multifrequency Relay section of Dial Peer Configuration on Voice Gateway Routers.
For information on configuring DTMF that is specific to call control protocols, see the following:
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CODECs
The following CODECs are supported by the Cisco 880 series voice gateway routers.
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For information on CODECs, see the following:
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SCCP-Controlled Analog Ports with Supplementary Features
Cisco 880 series voice gateway ISRs support the Cisco Skinny Client Control Protocol (SCCP) that supplies supplementary features on analog voice ports that are controlled by Cisco Unified Communications Manager or by a Cisco Unified Communications Manager Express system. Supported features include:
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For more information on the features supported and their configuration, see SCCP Controlled Analog (FXS) Ports with Supplementary Features in Cisco IOS Gateways.
Fax Services
The Cisco 880 series voice gateway ISRs support the following fax services:
Fax Pass-Through
Fax Pass-Through is the simplest way of transmitting faxes over IP, although it is not as reliable as Cisco Fax Relay. See the Configuring Fax Pass-Through chapter of the Cisco IOS Fax and Modem Services over IP Application Guide for more information.
Cisco Fax Relay
Cisco Fax Relay is a Cisco proprietary fax method that is turned on by default. Cisco Fax Relay allows the relay of a T.30 modulated signal across IP gateways in real-time on H.323 or SIP networks. See the Configuring Cisco Fax Relay chapter of the Cisco IOS Fax and Modem Services over IP Application Guide for more information.
T.37 Store-and-Forward Fax
The T.37 Store-and-Forward Fax mechanism allows a gateway to store and forward fax messages on H.323 or SIP networks. See the Configuring T.37 Store-and-Forward Fax chapter of the Cisco IOS Fax and Modem Services over IP Application Guide for more information.
T.38 Fax Relay
The T.38 Fax Relay provides an ITU-standard mechanism for real-time relay of fax signals. Gateway-controlled T.38 Fax Relay is available on MGCP networks. See the Configuring T.38 Fax Relay chapter of the Cisco IOS Fax and Modem Services over IP Application Guide for more information.
Unified Survival Remote Site Telephony
Cisco 880 Series voice gateway ISRs with Unified Survival Remote Site Telephony (SRST) include the following:
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Unified SRST automatically detects a failure in the network and initializes the process of auto configuring the router. Unified SRST provides redundancy for the IP and FXS phones to ensure that the telephone system remains operational.
All the IP phones and analog phones connected to a telecommuter site are controlled by the headquarters office call control system, which uses Cisco Unified Communications Manager. During a WAN failure, the telecommuter router allows all the phones to reregister to the headquarter in SRST mode, allowing all inbound and outbound dialing to be routed off to the PSTN (on a backup Foreign Exchange Office (FXO) or BRI port). Upon restoration of WAN connectivity, the system automatically returns communication to the primary Cisco Unified Communications Manager cluster.
Direct Inward Dialing (DID) is supported on the Cisco 880 series SRST voice gateway ISRs.
For general Unified SRST information, see the Cisco Unified SRST System Administrator Guide. Cisco Unified SRST is described in the Overview chapter.
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For SIP-specific SRST information, see the Cisco Unified SIP SRST System Administrator Guide. To configure SIP SRST features, see the Cisco Unified SIP SRST 4.1 chapter.
Verification of Voice Configuration
Use the following procedures to verify voice port configurations:
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To verify, monitor, and maintain SRST, see Monitoring and Maintaining Cisco Unified SRST.