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Unified Communications
Featured Article
UC Mag
Richard "Zippy" Grigonis
Executive Editor,

IP Communication Group

SIP and Unified Communications

SIP (Session Initiation Protocol) is an end-to-end application layer signaling/call control protocol that first appeared in 1999 and is now defined by 150+ RFC (Request For Comments) documents by the IETF (Internet Engineering Task Force). SIP can control real-time packetized multimedia sessions between participants over an IP network, operating independently of any underlying transport protocol such as IP, RTP and UDP. It remains the premier VoIP call control standard in both wireline and wireless 3G networks, as defined by the Third Generation Partnership Projects (3GPP). CableLabs incorporated SIP into the PacketCable 2.0 standard in 2005, and SIP is the principal protocol in IMS (IP-based Multimedia Subsystem), common service architecture for fixed-line and wireless networks now slowly undergoing deployment worldwide.

 




Clearly, any UC system worth its salt must heavily rely on SIP, since it facilitates "one number" addressing, intelligent routing based on device modalities/characteristics, and session transfers, which is the ability to switch live media connections between devices. Moreover, as IP network connections proliferate, businesses are now pondering whether they should adopt so-called SIP trunking, which enables them to dispose of their in-house gateways and BRI and PRI lines connecting their IP phone systems to the old PSTN, instead relying on their Internet Telephony Service Provider (ITSP) to handle the gateway function. Provided that enough other businesses, suppliers, and partners are also IP-connected, money-saving direct connections over IP are now possible, as well the use of high-fidelity, wideband voice codecs. SIP trunking requires use of a PBX with a SIP-enabled trunk side, a SIP-capable enterprise edge device understanding SIP and an Internet telephony or SIP trunking service provider.

 

One former impediment to SIP trunking (and VoIP in general) has been corporate firewalls and NAT (Network Address Translation) devices that can be found at the edge of virtually all business networks. Most firewalls don't support unsolicited incoming media flows, and NAT devices (or software) allow client devices on the corporate LAN to "hide" from the outside world via private IP addresses and ports inserted into packets, making them non-routable in public networks.

 

As it happens, companies such as Ingate Systems AB of Stockholm, Sweden, provide the functionality to resolve this problem. In Ingate's case, their SIP Trunking software module works in conjunction with an Ingate Firewall or SIParator to solve any NAT traversal issues encountered by businesses using a SIP trunk. Together, they control both incoming and outgoing communications and route the communication to intended users.

 

Ingate SIP Trunking can direct international calls to national, or local, PSTN lines within the country being called. Businesses can use multiple service providers in a least cost routing rules scheme, and switch between the providers depending on which offers the best possible rates at that particular time, day or location. Long distance calls cost the same as a local call, reducing expenses for businesses as well as their customers, partners, etc., attempting to reach salespeople, for example. Multiple SIP trunk support also allows for redundancy. (If a connection to one ITSP goes down, Ingate can immediately transfer the traffic to another ITSP.)

 

Ingate SIP Trunking also supports ENUM ("Telephone NUmber Mapping" or "Electronic NUMbering"), which can automatically look up phone numbers to determine whether they match a known SIP address, allowing the call to be completed over the Internet instead of transferring it out to the PSTN. No traffic is sent over the PSTN, so ENUM provides yet another way of saving money if a business communicates with other firms using SIP.

 

Big Vendors Adopt IP, and SIP Joins the Ride

 

Since UC and SIP encompass whole suites and collections of applications, many former point product providers have repositioned themselves as one-stop total solutions providers. At Alcatel-Lucent, for example, they've greatly enhanced and revamped many of their switch and UC products. Their OmniPCX Enterprise platform is an integrated, interactive communications solution for medium-sized businesses and large corporations. The solution combines traditional telephone functions with support for Internet-based telephony and multimedia communication. OmniPCX Enterprise provides a suite of UC applications, including a web softphone, along with unified messaging and personal assistant applications. Based on a single software architecture, the OmniPCX Enterprise platform is compatible with multiple operating systems and integrates easily into your data infrastructure.

 

The Linux-based communication server has an integrated SIP proxy and SIP gateway which enables SIP devices to be assigned a directory number and become part of the enterprise dial plan, so SIP users can be placed in the OmniPCX directory. OmniPCX Enterprise has LDAP directory capabilities, so SIP users also become part of the larger corporate directory infrastructure and can be dialed by name. These same SIP users can appear as extensions within the OmniPCX and be assigned a class of service, linked to call accounting and tied into voice mail. Non-SIP extensions can be routed to SIP devices, including client, gateway, and proxy, using Automatic Route Selection (ARS). Alcatel-Lucent supports third party SIP devices through the OmniPCX OmniPCX Enterprise's gateway and proxy servers.

 

The OmniPCX Enterprise SIP gateway provides connections to analog phones, digital phones and IP telephones as well as PSTN access. If the SIP devices support features such as caller or called party ID, hold, transfer, forwarding, etc., the gateway ensures these are supported on the OmniPCX Enterprise as well. Again, SIP supports an evolving set of PBX features. The gateway can provide a message waiting indication to the SIP device when voice mail has been delivered.

 

The OmniPCX Enterprise SIP proxy server provides dynamic location and routing for SIP communications. The proxy implements parallel forking so a user can be called simultaneously on several SIP devices. The OmniPCX SIP registrar dynamically updates the location database, when it receives notification that users are online. Because the proxy server "speaks" with the enterprise's Domain Name Server (DNS), it handles connections to other SIP proxies in the same or different domain. The SIP proxy also handles TCP and UDP network transport.

 

Over at Avaya, their "Unified Communications All Inclusive package" includes such things as the Avaya Communication Manager 5.0 IP Telephony software that leverages SIP to enhance connectivity, thus bringing about reduced costs for trunking and connecting systems, faster deployment of new capabilities and increased options for linking communications and business systems. Also, Avaya's one-X Communicator next-gen softphone supports both the older H.323 protocol as well as SIP audio and video, and providing access to all communications media in a single interface (and the Avaya one-X Portal delivers web-based telephony, messaging, contacts and conferencing through any public or private Windows, Mac and or Linux-based computer).

 

Avaya's UC All Inclusive package is automatically included for new customers and those upgrading to Avaya Communication Manager 5.0 Enterprise Edition for no additional cost. For customers of Communication Manager 5.0 Standard Edition, the package is available for US$50 per user license.

 

And although Cisco Systems may have initially built its reputation with its proprietary Skinny Client Control Protocol (SCCP) used between the Cisco Call Manager and Cisco VoIP phones, it has also now implemented SIP natively into its UC systems, so that customers may choose SIP, SCCP or a combination of both, with no incremental hardware required. The Cisco Unified Communications Manager (CCM) can be deployed with SIP and Presence. The Cisco Unified Communications Manager (CUCM) 6.0 and now 7.0 supports SIP endpoints and SIP trunking, and the Cisco Unified Communications Manager can be integrated with Cisco Unified Presence (CUP), a component that collects information about a user's availability status and communications capabilities, including whether or not that user is using a communications device - such as a phone or Cisco Unified Video Advantage - or if that user has web collaboration or videoconferencing enabled on their system. It's possible to configure CUCM for Presence and create a SIP trunk for connection to the CUP server, using SIP and the SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE). The addition of SIP enabled Cisco to invite third-party developers to develop new add-on applications (such as those for vertical or horizontal industries) for Cisco UC systems. Cisco Communications Manager and Cisco Unified Presence also enable third-party SIP-based hardware and integration with third-party presence servers.

 

Intelligent SIP

 

Interactive Intelligence, one of the world's great providers of unified IP business communications solutions for contact center automation, enterprise IP telephony, and enterprise messaging, relies heavily on SIP technology, such as their Interaction Media Server, SIP Proxy, and related equipment. To be more specific, their Interaction Gateway works with SIP-based telephony systems, especially when paired with Interactive Intelligence's legendary Customer Interaction Center (CIC) IP contact center/enterprise platform or Vonexus Enterprise Interaction Center (EIC) IP PBX.

 

As for the Interaction SIP Proxy, it's a comprehensive SIP proxy for load balancing and business continuity management that's easy to install and use. You just load the Interaction SIP Proxy software on an inexpensive Windows-based machine, and in a short time you have a sophisticated SIP Proxy server up and running.

 

To increase the performance of a SIP-supported CIC or Vonexus EIC solution, you can move the system's audio recording and processing to a dedicated, reliable Interaction Media Server, which leverages next-gen IONâ„¢ open technology from Interactive Intelligence.

 

SIP Has Become a Gulp

 

Some may say that the SIP protocol rode in on the coattails of VoIP, but, in fact, if SIP didn't exist we would have had to invent something like it anyway. It will continue to exist at the heart of all advanced IP communications, particularly in the many dimensions of unified communications.

 







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