SIP trunking is a Voice over Internet Protocol (VoIP) and streaming media service based on the Session Initiation Protocol (SIP)[1] by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and Unified Communications facilities.[2] Most Unified Communications software applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard

SIP trunks in Jordan

TelecomsXChange signes a 5 year contract with a Jordanian licensed operator (Alkaram w Al Jood) to provide onsite consultancy and support for local financial institutions, Education institutions and small and medium sized businesses. Once the SIP trunking equipments is installed & configured it will be pointed out of the box to TelecomsXChange platform which will give you access to the lowest international rates in the market instantly.

You may contact us immediately at for any assistance or questions in regards to SIP trunking for your business, our remote and local partners will work with you to provide you a complete demo and understanding of every step that will be taken to power your contact center with the next generation communication capabilities and  lowering your costs by thousands of dollars a month.

Our Advantages:

  1. Bring your own internet provider
  2. Unlimited channels, Pay only for what you use
  3. High priority fault handling
  4. Multiple geographical redundancies 
  5. 24 hours setup time 
  6. No yearly commitments, cancel anytime for free
  7. Get rate decreases to your top destination as they happen in the international wholesale market
  8. Easy on boarding process, usually takes 1-2 hours to get your account live
  9. No IP-PBX ? No problem, our team will guide you through what equipment you need to purchase to upgrade your old PBX to an IP-PBX.

SIP trunking features

A SIP Trunk is a logical connection between your IP PBX and TelecomsXChange servers that allows voice over IP traffic to be exchanged between the two.

When a call is placed from an internal phone to an external number, the PBX sends the necessary information to TelecomsXChange servers (SIP Trunk provider) who establishes the call to the dialed number and acts as an intermediary for the call. All signaling and voice traffic between the PBX and TelecomsXChange are exchanged using SIP and RTP protocol packets over the IP network.

There are three components necessary to successfully deploy SIP trunks: a PBX with a SIP-enabled trunk side, an enterprise edge device understanding SIP and an Internet telephony or SIP trunking service provider, TelecomsXChange.


In most cases the PBX is an IP-based PBX, communicating with all endpoints over IP, but it may just as well be a traditional digital or analog PBX. The sole requirement is that an interface for SIP trunking connectivity is available.

The enterprise border element

The PBX on the LAN connects to the Mada via the enterprise border element. The enterprise edge component can either be a firewall with complete support for SIP or an edge device connected to the firewall, handling the traversal of the SIP traffic. 

TelecomsXChange (An award winning open trading platform for next generation Voice carriers) provides connectivity to the PSTN (Public Switched Telephone Network) for communication with mobile and fixed phones via over 300 carriers wold-wide.