Session Initiation Protocol (SIP) has become the leading call management protocol for setting up, conducting and terminating real-time IP media calls within the corporate perimeter, between distributed IP telephony networks, and between a VoIP network and a PBX. IP media supported by SIP can include VoIP calls, instant messaging (IM), multimedia conferences, user presence information, Enhanced 9-1-1 (E9-1-1) emergency calls, and other communications. SIP has also become the dominant signaling protocol for connecting VoIP networks to the Public Switched Telephone Network (PSTN), which can be implemented via an IP-PSTN trunk to an Internet Telephony Service Provider (ITSP).
Cisco offers a variety of SIP-only VoIP and video IP phones and currently recommends SIP over other supported signaling protocols like H.323 for trunking because of SIP's richer feature set. In Cisco's portfolio of unified communications technologies, the Cisco Unified Border Element (CUBE) has superseded traditional TDM gateways as the preferred method for connecting to the PSTN. CUBE offers all the key features required for a SIP PSTN trunk including demarcation, call admission control, session management, interworking, QoS, and security checks.
Progent's Cisco-certified CCIE Collaboration consultants offer online and onsite expertise to help organizations of any size to create and manage SIP infrastructure solutions based on Cisco technology. Services offered by Progent include Cisco Unified Communications Manager (CUCM) consulting, Cisco VoIP phone and Cisco video IP phone configurations and troubleshooting, SIP trunking support, and CUBE planning and integration. Progent can help you migrate from H.323 trunks and gatekeeper-based CUCM trunks to SIP trunks, configure Cisco ISR G2 routers with CUBE support, and upgrade from an older PBX to a modern IP PBX system. Progent offers as-needed expertise to help you through stubborn technical roadblocks or Progent can provide comprehensive project management outsourcing or co-sourcing services to make sure you complete your SIP integration initiative on time and on budget.
SIP PSTN Trunking
In Cisco's collaboration architecture, a trunk is a Cisco Unified CM (CUCM) or Unified Communications Manager Express (CME) communications channel that allows CUCM to interface with other servers, which can be local or external. Trunks allow CUCM to initiate or receive VoIP and IP video calls, deliver real-time event data such as Presence, and provide other collaboration services. SIP trunks are popularly associated with connecting to an IP-PSTN service provider, but SIP trunks have other important uses. Although CUCM continues to support H.323 trunks, Cisco cautions that the feature gap between SIP and H.323 is widening. For example, SIP trunks support TLS signaling encryption, H.264 video with BFCP-based presentation sharing, RSVP-based QoS and call admission control, support for the "+" character, protocol modification scripts for interoperability, and URI based calls. In addition, Cisco's SIP trunks can support IPv4 only, IPv6 only, or Dual Stack (IPv4 and IPv6) ANAT-enabled trunks.
SIP trunks simplify the design and management of Cisco Collaboration Systems networks
SIP devices that can be connected via SIP trunks include gateways, Cisco Unified CM Session Management Edition, SIP proxies, Unified Communications applications, and other CUCM clusters. Current releases of CUCM support a growing array of SIP trunk and call routing features such as:
SIP Trunk Deployment Models
- SIP trunk calls can be placed or answered on any subscriber node in the CUCM cluster
- 16 destination IP addresses are supported per trunk, reducing the need for multiple trunks
- SIP OPTIONS ping keep-alives let you track the current state of the trunk's destinations
- Support for Delayed Offer, Early Offer, and Best Effort Early Offer provides deployment flexibility
- Codec preference lists streamline configuration of SIP trunks between CUCM and Cisco Unified Border Element (CUBE)
- Normalization and transparency scripts simplify interoperability with third-party PBXs, apps, and IP PSTN service providers
- H.264 Video with BFCP and Far End Camera Control enhances collaboration
IP-PSTN SIP trunks can be configured using a centralized, distributed, or hybrid deployment architecture. In the centralized model, a SIP trunk or set of trunks connect all your network sites and remote users to an IP PSTN service provider via a shared logical interface using a Session Border Controller (SBC) such as the Cisco Unified Border Element (CUBE). You can use CUCM Session Management Edition (SME) to aggregate PSTN connections to a common PSTN trunk. The centralized model minimizes equipment costs and reduces deployment and management complexity by providing advantages that include the reduced need for branch-based IP PSTN trunks, a centralized dial plan and corresponding reachability information, a single aggregation point for multiple PBX systems, plus centralized applications like conferencing and Cisco Unity voice mail. A downside of the centralized model is that all the media and signaling traffic for PSTN calls must travel across the IP wide area network, which requires a high availability and high performance infrastructure.
Centralized SIP PSTN trunks are economical and simplify deployment and management
In the distributed model, each branch of a multi-site network can have its own local SIP trunk to the IP PSTN service provider, so media and signaling traffic goes through a local CUBE session border controller. The distributed trunk deployment model allows local hand-off of media and offers superior number portability from local providers. In addition, because centralized PSTN trunks can occasionally experience wide area network service breakdowns or traffic overloads, it is good practice for remote sites to have local PSTN gateways capable of placing emergency 911 calls.
Distributed SIP PSTN trunks can be more resilient and allow better number portability
A hybrid connectivity model for SIP trunks lets you aggregate some branches for shared IP PSTN connectivity while allowing other branches to have their own local IP PSTN trunk. The hybrid model can also provide trunks from each CUCM cluster within a multi-cluster deployment. This model can be more complex to deploy and manage but offers some of the functionality benefits of both the centralized and distributed deployment architectures. Progent's CCIE Collaboration-certified Cisco infrastructure consultants can help your organization to determine which SIP PSTN trunking architecture best addresses your IP telephony needs and Progent can assist you to design, deploy and maintain a SIP PSTN trunk solution that delivers maximum business value.
Cisco Unified Border Element (CUBE)
Cisco Unified Border Element (CUBE) is a licensed Cisco IOS application that uses a special IOS feature set to route calls between VoIP dial peers. A CUBE-enabled device can function as a session border controller (SBC) that makes it easy and affordable for businesses of any size to implement a SIP PSTN trunk. CUBE can run on a Cisco 2900 Series, 3900 Series, or 4000 Series Integrated Service Router (ISR), on a Cisco Aggregation Services Router (ASR), or can be deployed virtually to run on Cisco UCS servers. CUBE can scale to support from 4 to 64,000 concurrent voice calls and can provide in-box redundancy (on ASR routers only) or box-to-box failover for high availability.
CUBE can provide secure voice and video connectivity between two separate VoIP networks
Features and services provided by CUBE include:
For high availability, CUBE can be configured to provide full stateful failover for active SIP-to-SIP calls, preserving both media and signaling session information after switchover. For local and geographical redundancies you can configure high-availability CUBE clusters on ISR G2 and ASR routers. You can implement load balancing by using Cisco Unified SIP Proxy (CUSP) or a Service Provider (SP) call agent.
- Address and port translations for privacy and topology hiding
- SIP Delayed Offer to Early Offer conversion
- Interworking between SIP and H.323 protocols for easy migration to SIP
- Normalization to facilitate interoperability with IP PBXs and service providers
- DTMF translation, fax, transcoding, transrating, volume and gain control
- Granular call admission control
- Extensive security including SIP malformed packet detection, toll fraud protection, plus signal and media encryption
- Identity Header Interworking with Internet telephony service providers (ITSPs)
- Quality of Service (QoS) and bandwidth management with RSVP and codec filtering
- Concurrent connections to SIP trunks from multiple ITSPs
- URI routing using GDPR route-strings to match dial peers
- Domain-based routing
- Media forking or API capabilities to record voice or video calls
- Interoperability with Microsoft Lync and Skype for Business
- IP Phone registration to CUCM without requiring VPN
- Billing statistics and call detail record (CDR) collection and normalization
CUBE-enabled routers can be deployed in failover clusters with load balancing
Progent can help you select an appropriate CUBE solution based on the current and projected capacity requirements for your SIP PSTN trunk, configure ISR G2 routers, ASR routers and CUSP proxy servers for high availability SIP trunking, and work with your IP PSTN service provider to make sure the IP telephony features you need are properly configured and supported.
How Progent Can Help You with Your SIP Infrastructure and CUBE Integration
IP telephony is evolving quickly and few small or midsize organizations have in-house IT staffs with the time and the depth of experience to stay on top of this critical field. Legacy solutions such as outdated PBXs or obsolete trunk technology are common because many businesses are understandably reluctant to disturb their vital communications environments until pivotal products reach end-of-life and end-of-support or key services become unavailable. Progent's Cisco-certified collaboration consultants are seasoned experts at planning, integrating, managing and troubleshooting Cisco's SIP infrastructure and CUBE IP PSTN trunks and can provide affordable online or onsite support to help companies migrate efficiently to a modern, cost-effective IP telephony solution that delivers fast return on investment. In addition, Progent's certified Microsoft collaboration consultants can help you configure connections between Cisco CUBE and Microsoft's VoIP and IP video applications such as Skype for Business Server or Lync Server for connectivity to a SIP PSTN trunk.
Progent's GIAC certified information assurance consultants can help you configure the security options available with SIP technology such as signal and media encryption, and Progent can help you validate the compliance of your IP telephony environment with industry and regulatory requirements. Progent can help you design and deploy Cisco's high-availability technology for SIP trunks through CUBE clustering and load balancing, and Progent's disaster recovery and business continuity (DR/BC) planning consultants can help you create, test and document a disaster recovery plan to maximize the survivability and expedite the recovery of your critical communications resources.
SIP integration solutions involve an imposing collection of elements including physical and virtual servers, routers, firewalls, switches, proxies, management and application software, VoIP and video IP endpoints, debugging tools, multiple communication protocols, and IP carriers with varying levels of flexibility and support. As a consequence of this complexity, even seemingly minor upgrades can ripple through your IP telephony ecosystem to cause major disruptions. Progent has the experience to anticipate, identify and resolve most issues quickly and affordably via online support, and Progent has the ability to escalate support services to world-class infrastructure engineers. Progent's SIP experts also have the background to coordinate efficiently with Internet Telephony Service Providers to integrate SIP trunks, which is one of the most common technical challenges encountered by small and midsize business trying to upgrade or migrate their IP telephony environments.
Typical services provided by Progent's consultants for creating SIP trunking solutions include:
Contact Progent for Consulting Support for Cisco SIP Trunking Solutions
- Design, review, revise and add dial plans, NAT rules and access control lists (ACLs)
- Resolve IP phone registration, IP addressing and SIP authentication issues
- Add SIP user agents (UAs) and configure SIP-UA commands
- Troubleshoot DID issues
- Migrate from PRI gateways to SIP trunks
- Integrate CUBE with Lync and Skype for Business
- Review and update route groups, route lists, port forwarding rules, and partitions
- Set up Cisco SIP conference phones
- Add translations and set up dial-peers
- Debug Cisco and third-party firewall configurations and SIP PBXs
- Configure SIP failover solutions
- Set up and troubleshoot Cisco ISR G2 and ASR routers for SIP
- Configure Multiprotocol Label Switching (MPLS)
- Configure CUCM with SIP trunks
- Design clean cutover plans to new SIP trunks
- Troubleshoot QoS and performance problems and identify carrier bandwidth issues
- Resolve licensing issues
- Set up Proxy Servers
- Update External DNS
- Troubleshoot using RTMT logs
- Configure SIP Application Layer Gateways (ALGs)
- Create normalization scripts for SIP PSTN carriers
To talk to Progent about consulting services for Cisco SIP trunking technology, call 1-800-993-9400 or visit Contact Progent.
To see additional details about Progent's engineering support for Cisco technology, select a topic:
If you wish to contact Progent about technical support for Cisco networking, phone 1-800-993-9400 or go to Contact Progent.