Cisco Unified SIP Proxy [Cisco Network Modules]

Product Overview

The Cisco^® Unified SIP Proxy (USP) is a high-performance, highly available Session Initiation Protocol (SIP) server for centralized routing and SIP signaling normalization. By forwarding requests between call-control domains, the Cisco Unified SIP Proxy provides the means for routing sessions within enterprise and service provider networks. The application is delivered in a network module form factor on Cisco 2900, 3800, and 3900 Series Integrated Services Routers.

Cisco Unified SIP Proxy aggregates SIP elements and enables application of highly developed routing rules. These rules enable greater control, management, and flexibility of SIP networks. Cisco Unified SIP Proxy simplifies large Cisco Unified Communications Manager, Cisco Unified Communications Manager Express (CME), Cisco Unified Border Element, and Cisco Unified Customer Voice Portal (CVP) deployments as well as other Cisco and multivendor deployments. Cisco Unified SIP Proxy provides great scalability, and network design can provide high availability (Figure 1).

Figure 1. Cisco Unified SIP Proxy Network Module

Applications

Cisco Unified SIP Proxy routes among SIP elements and helps enable a broad range of unified communication services.

Cisco Unified Communications Manager and Cisco Unified Communications Manager Express SIP Aggregation

Management of SIP dial peers across midsize and large Cisco Unified Communications Manager and Cisco Unified Communications Manager Express networks presents a challenge. As opposed to a full mesh, dial peers can be pointed to Cisco Unified SIP Proxy, which provides a central route point. This process also simplifies the addition and removal of new call-processing agents. If a call-processing agent is unavailable, alternate routing and recovery can be provided. You can apply dial normalization and load balancing as needed (Figure 2).

Figure 2. Cisco Unified Communications Manager and Cisco Unified Communications Manager Express SIP Aggregation

Cisco Unified Border Element Scalability and Load Balancing

Cisco Unified SIP Proxy provides a central route point for management of multiple Cisco Unified Border Elements. You can establish logical separations and use a single Cisco Unified SIP Proxy for either or both ingress and egress traffic. You can apply load balancing and rule-based routing, and provide interfacing to the SIP trunk signal normalization where needed (Figure 3).

If a Cisco Unified Border Element is unavailable, Cisco Unified SIP Proxy can intelligently reroute to an alternate Cisco Unified Border Element. When the Cisco Unified Border Element returns to service, Cisco Unified SIP Proxy resumes sending traffic to the Cisco Unified Border Element. This design enables need-based growth of the service provider interconnect and also avoids risk associated with a single point of failure for the border element.

Figure 3. Cisco Unified Border Element Scalability and Load Balancing

SIP Trunk for Contact Center

Whether for inbound or outbound traffic, Cisco Unified SIP Proxy enables routing and management across contact center time-division multiplexing (TDM) and IP trunks. Routing is provided across gateways connecting outside the network as well as across multiple Cisco Unified Customer Voice Portals. If a Cisco Unified Customer Voice Portal or gateway is unavailable, Cisco Unified SIP Proxy can intelligently reroute to an alternate Cisco Unified Customer Voice Portal or gateway. When the Cisco Unified Customer Voice Portal or gateway returns to service, Cisco Unified SIP Proxy resumes sending traffic to the Cisco Unified Customer Voice Portal or gateway. You can apply load balancing and rule-based routing, and provide interfacing to the SIP trunk signal normalization where needed (Figure 4).

Figure 4. SIP Trunk for Contact Center

Cisco Unity PBX IP Media Gateway Integration

PBX IP Media gateways (PIMGs) are used to connect TDM-based private branch exchanges (PBXs) into Cisco Unity^® voice messaging systems. Placement of Cisco Unified SIP Proxy in front of the Cisco Unity application enables PIMGs to share Cisco Unity ports, in turn enabling scalability of hybrid TDM PBX and IP messaging deployments (Figure 5).

Figure 5. Cisco Unity PBX IP Media Gateway Integration

Service Provider SIP Interconnect Services

For interconnection among service providers, Cisco Unified SIP Proxy enables normalization of dial strings and SIP signaling variants. Cisco Unified SIP Proxy also provides routing and load balancing among SIP elements, including Cisco Unified Border Elements (Figure 6).

Figure 6. Service Provider SIP Interconnect

Product Architecture

Cisco Unified SIP Proxy Call Processing

The Cisco Unified SIP Proxy is a call and dialog stateless SIP proxy. Media flows around the proxy and signaling go through the proxy. The proxy can also modify SIP headers (normalization). Routing and normalization is determined based on administrator-configured policies. Policies are selected based on triggers, administrator-configured conditions that are matched based on information in the SIP message.

As SIP messages come into the proxy, a determination is made as to whether any prenormalization policies need to be applied. Following prenormalization, new triggers are used to determine application of routing policies. A further series of triggers provides for further header modifications; for example, postnormalization policies after the routing decision has been made. In cases where policy is not asserted, the proxy provides for pass-through of the SIP message (Figure 7).

Figure 7. Cisco Unified SIP Proxy Call-Processing Model

You can apply distinct rules to groups of requests to create independent "virtualized proxies" within a single Cisco Unified SIP Proxy. The rules are highly flexible and scalable to form routing or normalization policies.

You can deploy Cisco Unified SIP Proxy in redundant network designs. Redundant Cisco Unified SIP Proxies are deployed in an active-active mode. If one Cisco Unified SIP Proxy fails, another Cisco Unified SIP Proxy can assume the load of the failed one. You can place two Cisco Unified SIP Proxies in a single router or enable redundancy across multiple routers, even taking advantage of different geographic locations thereby avoiding the risk of a single location failure.

Hierarchical designs provide for high scalability. Cisco Unified SIP Proxies might be placed regionally. You can form multiple Cisco Unified SIP Proxies as a cluster with a higher-order Cisco Unified SIP Proxy.

Features

• Proxy for SIP unified communication signaling

• Signaling support: Voice, video, fax, physical terminal line (tty), modem, caller ID, caller name, updates, transfer, forward, hold, conference, status, message waiting indicator (MWI), dual tone multifrequency (DTMF) relay, and SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) (presence)

• Address resolution (Domain Name System [DNS]: Type A and SRV and Type NAPTR)

– Domain name resolution based on RFC 3263, Locating SIP Servers

• TCP, User Datagram Protocol (UDP), and Transport Layer Security (TLS)

• Available as a network module on Cisco 2900, 3800, and 3900 Integrated Services Routers

– Minimal performance effect on the router allows for concurrent router applications.

Routing

• Routing based on policy

• Configurable multistep routing policies with route table lookup

– Configurable match rules (for example, longest prefix, exact match, and fixed-length match)

– Configurable keys selected from the SIP request: Remote address, local address, request uniform resource identifier (URI), P-Asserted-Identity (caller ID), diversion, remote-party ID, To, and From; within these headers Cisco Unified SIP Proxy can select the user, host, port, domain, phone number, URI, carrier codes, and location routing numbers

– Configurable key modifiers (for example, case insensitivity, ignore plus, ignore display characters, etc.)

– Numerous routing decisions: Forward to a single route, forward to a route group, reject, and chain to another route policy

• Table-based routing for mapping of requests to destinations

– Support for large number of routes in a table (10,000+)

– Routes populated through command-line interface or upload of a route file

• Example routing scenarios

– URI-based routing (number and name)

– Call block between specified sources and destinations, including policy-based transit routing (policy may require certain calls to either avoid or to take certain routes)

– Class of restriction

– Translation of on-net to off-net dial plans (including public switched telephone network [PSTN] and IP-IP); simplifies network management, eliminating the need to configure translations in each call agent

• Percentage and weight-based routing

– Load balancing among downstream elements based on preset weight

– Priority values assignable for routing of selected calls; also enables configuration for least-cost routing

• Time policy routing

– Time(s) in a day, day(s) in a week, day(s) in a month, and month(s) in a year

• Ability to form downstream elements into a single logical group for load balancing and failover

• SIP element health management and monitoring

– Rerouting around unavailable SIP elements

– Ping for service availability and restoration of routing when unavailable SIP element is restored

• Rerouting based on redirect responses (Routing policy and postnormalization applies to the new destination specified in the contact header of the redirect response, and provides for sequential forking.)

• Transport protocol conversion: TCP, UDP, TLS (For example, an incoming call received over UDP can be forwarded to a destination over TLS.)

• Configurable record-routing (on/off)

• Global unique caller ID pass-through

Normalization

• Normalization of SIP headers based on configurable policy

– Ability to add, remove, or update headers and header parameters

– Ability to update URI components such as user, domain, and host and ability to add, remove, and update URI parameters

– Digit manipulation

– Address manipulation

– TEL URI <=> SIP URI conversion

– Domain conversions

– Regular-expression processing

• Construction of multistep normalization policies

• Pre- and postnormalization

– Prenormalization prior to proxy application of routing rules (for example, applied to message coming into the proxy)

– Postnormalization after proxy application of routing rules (for example, applied to message going out from the proxy)

Rules-Based Selection of Routing and Normalization Policies

• Rich set of configurable rules

– SIP message type (for example, request and response)

– SIP method: INVITE, UPDATE, REFER, PRACK, BYE, SUBSCRIBE, NOTIFY, unsolicited NOTIFY, MESSAGE, PUBLISH, REGISTER, INFO, OPTIONS, and any custom or future SIP extensions

– Request-URI: User, host, phone number, etc.

– Local and remote IP, port, and protocol of the received SIP message

– Network name of the incoming and outgoing request (A network is a set of SIP listening points.)

– Transport protocol

– Regular expression match on any SIP header

– Time policy check

– SIP response code

– Middialog message check

• Flexible rule management

– Ability to arrange rules in groups and prioritize them

– Boolean logic (and/or)

Security and Privacy

• TLS (bidirectional)

• Through-header stripping (for topology hiding)

• User privacy (RFC 3325 P-Asserted ID: Removes P-Asserted ID when receiving a message from an element not configured as trusted, and removes P-Asserted ID and Privacy header when forwarding a message to an element not configured as trusted)

Network Design

• Multiple IP addresses (up to eight) to provide for flexible configuration and network topology design; you can group IP addresses to form networks and apply rules on these networks

• Multiple SIP listening points; each listen point can have configurable port

• "Virtualized proxies" with multiple independent routing and normalization processing in a single server

• Redundancy through clustered network design for high availability

– Clusters addressed as Fully Qualified Domain Names (FQDNs). Domain Name System (DNS) resolution via Service (SRV) record

– Virtual IP addressing using Hot Standby Router Protocol (HSRP)

• Very high scalability with clustering of multiple Cisco Unified SIP Proxies

– Hierarchical and peer requests among clustered Cisco Unified SIP Proxies

– Up to two Cisco Unified SIP Proxy network modules within the same router and / or multiple Cisco Unified SIP Proxy network modules across different routers

Management

• RADIUS accounting for SIP events

• Flexible management through command-line interface

• Load or copy of preexisting configurations onto or off the Cisco Unified SIP Proxy module

• Graceful shutdown and restore, which allow for completion of transactions in process

• SIP message logging for call monitoring

• Trace logging for troubleshooting

• FTP access to Cisco Unified SIP Proxy for easy download of trace logs, SIP message logs, configuration files, and route files and upload of configuration files and route files

• SIP message metrics logging (peg counting); for example, count of incoming and outgoing messages over a period of time and logging to a file

Supported Standards as a SIP Proxy

• IETF RFC 2246: The TLS Protocol Version 1.0

• IETF RFC 2327 SDP: Session Description Protocol

• IETF RFC 2617 HTTP Authentication: Basic and Digest Access Authentication

• IETF RFC 2782: A DNS RR for specifying the location of services (DNS SRV)

• IETF RFC 2806: URLs for Telephone Calls

• IETF RFC 2976: The SIP INFO Method

• IETF RFC 3204: MIME media types for ISUP and QSIG Objects

• IETF RFC 3261 SIP: Session Initiation Protocol

• IETF RFC 3262: Reliability of Provisional Responses in the Session Initiation Protocol (SIP)

• IETF RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers

• IETF RFC 3264: An Offer/Answer Model with the Session Description Protocol (SDP)

• IETF RFC 3265: Session Initiation Protocol (SIP)-Specific Event Notification

• IETF RFC 3311: The Session Initiation Protocol (SIP) UPDATE Method

• IETF RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks

• IETF RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP)

• IETF RFC 3515: The Session Initiation Protocol (SIP) Refer Method

• IETF RFC 3665: Session Initiation Protocol (SIP) Basic Call Flow Examples

• IETF RFC 3666: Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows

• IETF RFC 3725: Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)

• IETF RFC 3842: A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)

• IETF RFC 3856: A Presence Event Package for the Session Initiation Protocol (SIP)

• IETF RFC 3891: The Session Initiation Protocol (SIP) "Replaces" Header

• IETF RFC 3892: The Session Initiation Protocol (SIP) Referred-By Mechanism

• IETF RFC 4480 RPID: Rich Presence Extensions to the Presence Information Data Format (PIDF)

• SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE)

Ordering

Cisco Unified SIP Proxy employs a counted feature license based on a maximum number of new incoming SIP requests per second. Requests that belong to an existing dialog, including SIP responses, are not counted.

To order, visit the Cisco Ordering Tool and refer to Table 1.

Table 1. Ordering Information

Cisco Unified SIP Proxy	Part Number
Cisco Unified SIP Proxy network module	NME-CUSP-522
Cisco Unified SIP Proxy base software	SCUSP-1.1-K9
License for 10 new incoming SIP requests/second	FL-CUSP-10
License for 30 new incoming SIP requests/second	FL-CUSP-30
License for 100* new incoming SIP requests/second	FL-CUSP-100
License upgrade 10 to 30 new incoming SIP requests/second	FL-CUSP-10UP30=
License upgrade 10 to 100 new incoming SIP requests/second	FL-CUSP-10UP100=
License upgrade 30 to 100* new incoming SIP requests/second	FL-CUSP-30UP100=
Network Module Adapter required on 2900 and 3900 Integrated Services Routers only	SM-NM-ADPTR

* The license for 100 new incoming SIP requests/second is not limited to 100. The 100 count is intended as guidance for many common applications. In some deployments substantially higher performance is expected. Also, in some cases the processing requests may be so substantial that this guidance is too high. Further details are provided in the "Performance" section.

The Cisco Unified SIP Proxy network module operates on the Cisco 2900, 3800, and 3900 Series Integrated Services Router platforms with any Cisco IOS /K9 image. You can employ multiple integrated services routing applications at the same time. However, Cisco Unified SIP Proxy may not co-reside in the same router when Cisco Unified Communications Manager Express or Cisco Unified Survivable Remote Site Telephony is configured for Skinny Client Control Protocol (SCCP) phones. Nor may Cisco Unified SIP Proxy co-reside in the same router with TDM gateways or configuration of H.323 dial peers (including Cisco Unified Border Element). Cisco Unified Communications Manager Express, Cisco Unified Survivable Remote Site Telephony, and Cisco Unified Border Element configured for SIP may co-reside in the same router. Other voice and router functions are also available for use in the same router with Cisco Unified SIP Proxy.

Performance

Performance is limited by both the number of incoming SIP requests specified in the feature license and module processing capability. The 100-count feature license is an exception. The 100-count feature license is not restrictive; module processing capability is the only limit to performance.

Table 2 describes maximum Cisco Unified SIP Proxy performance based on module processing capability alone.

Table 2. Performance Measured in the Number of New Call Attempts per Second

	Record Route ON		Record Route OFF
	UDP	TCP	UDP	TCP
Routing only*	200	140	480	440
Normalization with limited routing**	140	125	460	430
Cisco Unified Customer Voice Portal deployment***	250	225	500	450

* Measured using two route tables with 10,000 entries each; no normalization enabled

** Assumes normalization enabled for every SIP request; one route table with two entries

*** Performance is measured based on 1000 routes and no normalization.

Performance will vary depending on call flows. Performance will be lower when DNS lookup, SIP logging, or RADIUS logging services are enabled.

Hardware Specifications

Table 3. Hardware Specifications

Hardware part number	Cisco NME-CUSP-522-K9
Form factor	Enhanced network module (NME)
CPU	1.4-GHz Intel Pentium M
Memory (RAM)	2 GB
Storage	160-GB hard disk
Supported integrated services router platforms	Cisco 3800 Series Integrated Services Routers (Cisco 3825 and 3845 only)
Internal network interfaces	10/100/1000 Gigabit Ethernet connectivity to router backplane
Cisco IOS^® Software (on router)	Cisco IOS Software Release 12.4(22)T on the Cisco 3800 Integrated Service Routers Cisco IOS Software Release 15.0(1)M on the Cisco 2900 and 3900 Integrated Service Routers Supported using any of the Cisco IOS /K9images
Physical characteristics	Dimensions (H x W x D): 1.55 x 7.10 x 7.2 in. (3.9 x18.0 x 18.3 cm) Weight: 1.5 lb (0.7 kg) maximum
Operating environment	Operating temperature: 41 to 104°F (5 to 40°C) Nonoperating and storage temperature: -40 to 158°F (-40 to 70°C) Operating humidity: 5 to 85% (noncondensing) Operating altitude: -197 to 6000 ft (-60 to 1800m)
Safety	UL 60950-1, Safety of Information Technology Equipment-Safety-Part 1: General Requirements (USA); plastic materials that are exposed to the end user shall meet the requirements of fire enclosure (UL94V-1) as defined in UL 60950
EMC	Emission: • 47 CFR Part 15 Class A • CISPR22 Class A • EN300386 Class A • EN55022 Class A • EN61000-3-2 • EN61000-3-3 • SD/EMI (India) • KN22 (Korea) • VCCI Class I • AS/NZS CISPR 22 • Class A Immunity: • CISPR24 • EN300386 • EN50082-1 • EN55024 • SD/EMI (India) • KN22 (Korea) • EN61000-6-1

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For More Information

For more information about Cisco Unified SIP Proxy, contact your local Cisco account representative.

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