sipXecs is a new open source unified communications system offering voice, video, presence and IM. You could call it a SW based IP PBX or a SIP softswitch, but it can really do a lot more than what you are used to getting from a traditional PBX. In case you were wondering: sipXecs is definitely not based on Asterisk.

If you are in the market to replace your current communication system, you should look at sipXecs. sipXecs is a viable alternative to systems from well known vendors such as Cisco, Shoretel, Avaya, Nortel, Mitel, Siemens, NEC and also Microsoft OCS.

sipXecs is built for 32bit and 64bit x86 and Power PC (PPC) based platforms.

Basically you need a standard server (an old PC will do to try it), the sipXecs software, and some SIP phones (soft or hard). For trunk lines you can use SIP trunking or use a PSTN gateway.  sipXecs is SW only and does not require any special HW. It runs on the Linux operating system (Fedora, Red Hat, and SuSE). Installing a minimum of 1 GB of RAM would help.

sipXecs is based on a very different architecture and therefore it really does scale. The things that make it scalable are:

• sipXecs is a SIP message router based on a proxy based architecture. It is NOT a back-to-back user agent (B2BUA) design. This means that media does not go through the sipXecs server but travels on the LAN.
• sipXecs is a centrally managed distributed system. Individual functionality can be farmed out and run on dedicated hardware. This allows for scale AND redundancy.

The largest system supports 6,000 users. To be specific: ONE sipXecs system supports 6,000 phones and works as the primary phone system. It is NOT a loosely coupled collection of independent systems that have to be individually configured and managed. We are using a centrally managed high-availability system that consists of a load-balancing redundant call server.

sipXecs supports both and scales with both. The reason again is that media (voice and video) do not traverse the sipXecs server but stay on the LAN. sipXecs by default uses HD voice provided the phones support it. Video is supported between endpoints that support it.

There is no limit other than the traffic your LAN can support. Connections are setup between end-points directly from the beginning and sipXecs does not have to attempt to re-INVITE calls in an attempt to route them around the call server.

sipXecs does not care about codecs. Any codec the end-points involved in a conversation can use are acceptable. For calls into services provided by sipXecs, such as voicemail, auto-attendant, ACD, conferencing, etc., sipXecs supports different codecs. All components support G.711 of course. Most components support HD Voice based on G.722.

Yes, it does. It's like switching from a PSTN phone conversation to FM radio.  Audio becomes much more natural and easier to understand.sipXecs release 4.0 supports HD Voice not only between phones capable of HD Voice but also for calls into the auto-attendant and conferencing bridge. A new voicemail system is in the works that supports HD Voice.

Yes, sipXecs release 4.0 adds a very capable and scalable SIP trunking gateway. We think it is as good as commercial SBCs and offers significant scalability. In our load tests we ran 450 trunk calls simulteneously on a small desktop PC with the SIP trunking gateway only consuming 16.5% of the CPU capacity. In addition, sipXecs offers load-sharing redundancy for the SIP trunking gateway.

The sipXecs call server offers load-sharing redundancy. It is NOT a cold or warm standby system. If one of the sipXecs call servers goes down, no calls are interrupted and the user will not notice an outage. sipXecs release 4.0 supports two redundant call servers, centrally configured and managed. The limit to 2 call servers is a bit artoificial and mostly reflects limitations in our testing process. We are planning to open this up and allow many parallel load-sharing call servers, such as for survivable branch deployments. sipXecs release 4.0 does not offer redundancy for media services. However, the media services components can run on dedicated hardware and do not have to be co-located with a call server (however, they can be). Making individual media services redundant is part of the plan.

It is really simple. sipXecs does it all for you. All the phones and gateways are auto-discovered and auto-configured. sipXecs generates all the configuration profiles and makes them available on its server so that the phones and gateways can pick them up as they power up.  It only takes a few clicks to create a new user, assign a phone, and get it up and running. However, if you are an expert you can still go into the sipXecs admin console and tweak every parameter a phone or gateway has to offer. Never again will you have to login to the phone or gateway directly. All config is then centrally backed up. Phone config includes auto-discovery, general config, speed dials, BLF presence lists, directory, time and DST, and firmware upgrade.

sipXecs supports a long list of phones and gateways plug & play. However, not all phones and gateways are equally well tested. For a production system we recommend the following phones: Polycom Soundpoint and Soundstation, Nortel 1200, Counterpath Bria Professional, LG-Nortel 68xx. Best supported gateways are from Audiocodes. Patton gateways are known to work but need to be configured manually. We also know of users using Cisco gateways. A large number of additional phones are plug & play managed, including Aastra, Cisco, Linksys, Grandstream, Snom, IpDialog and some others.

Yes, SIPfoundry and the FreeSWITCH project cooperate. sipXecs uses FreeSWITCH as its media server for conferencing and auto-attendant. A FreeSWITCH based voicemail system will be added soon. sipXecs completely configures and manages FreeSWITCH so that from an admin perspective it is just a component of sipXecs. We choose FreeSWITCH because it performs very well, supports many different codecs, and is of solid quality.

sipXecs is often used as the reference implementation of a standards based SIP system. sipXecs is a native SIP call server that is throughout based on SIP. In particular this means that sipXecs uses SIP for all internal communication between components and also to realize feature interactions.

We believe SIP is the future. The architectural compromise is too big if you want to accommodate other protocols. It would mean that we would have to implement a proprietary core with channel drivers that support different protocols.  Because sipXecs is completely built using SIP it has a very solid and feature rich SIP implementation.

With release 4.0, sipXecs includes a complete remote worker solution that makes it unnecessary to deploy an additional SBC. sipXecs natively detects and compensates for far-end and near-end NATs. An optionally redundant media relay is used to anchor the media.

sipXecs supports presence using a resource list (RLS) based presence server. It is used to provide busy lamp field (BLF) presence on the phone.

sipXecs includes an ACD server that is capable of accommodating about 50 agents. It is best used for informal call centers, like IT help desks. In addition it can be used to handle inbound calls into support or sales, but you should not expect all the functionality a larger call center application will give you. It provides both historic and real-time statistics and it also offers a SOAP based Web Services API for real-time stats and agent presence.