VoIP - Services
Eurus provide VoIP (Voice over Internet Protocol) service and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. Some cost savings are due to utilizing a single network to carry voice and data, especially where users have existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to PSTN may have a cost that's borne by the VoIP user.
Eurus VOIP Network
It is a connectionless protocol in which packets can take different paths between the endpoints and all paths are shared by packets from different transmissions. This enables efficient allocation of network resources, as packets are routed on the paths with the least congestion. Header information makes sure that the packets reach their intended destinations and helps reconstruct the messages at the receiving end. To ensure QoS, however, all packets should use the same path. IP headers are large (20 bytes) as compared to the headers of Frame Relay frames (2 bytes) and of ATM cells (5 bytes). Headers therefore add a lot of overhead to IP traffic.
IP networks employ the same types of bandwidth-saving schemes as the Frame Relay network, including fragmentation, jitter buffering, prioritization, voice compression, silence suppression and echo canceling.
In-bound Services by Eurus
Inbound Services of Eurus gives our customers an attractive and cost effective means of keeping in contact with you. The vast range of features available with Eurus Inbound gives you the power to manage your inbound calls, monitor performance of your contact centres in real time, and change how, when and where your calls are answered.
Voice over IP (VoIP) service primarily a cheaper is a technology alternative to traditional landline phones. Now, it may also be a way for smaller companies to look a lot bigger in the global pond. We use the Internet-based architecture as a platform to provide inexpensive local telephone numbers and inbound telephony services to companies anywhere in the world.
- Toll Free
We provide an International toll free teleconference solution no matter where you are in the world is truly amazing. Our users can have Conference call over the phone in no time delay with ease and audio clarity. Our Toll Free leverages VoIP Toll Free services to leverage traditional telephone service capabilities while maintaining the control they need to create new products and services. With these services,customers can create and deliver applications faster, maximize call routing and back-end signaling efficiency, reduce maintenance and equipment costs, and increase their overall return on investment.
- US DID
(Direct Inward Dialing) and access numbers.
DID will connect you directly to the VoIP user while access numbers require you to input the extension number of the VoIP user. Stations on a private communications network are identified by an unique private network identification (ID) code or dial in direct (DID) number. Remote communications devices (e.g., cell phones, analog phones, etc.) are virtual client devices connected from a public communications network to the private communications network. One or more DID numbers are
shared DID numbers, dedicated for the virtual client devices. Calls from the virtual client devices may be placed to network stations by dialing shorthand (e.g., 5 or fewer digits) numbers. Each call has a dedicated DID number temporarily assigned to it for placing the call as an in network call. Once the call reaches its final destination, the shared DID number is released for temporary assignment to other virtual client device calls.
With our Universal International Free phone Number (UIFN) you can make inbound toll free calls outside of the Native country and extended call coverage areas. Once an ITFS or UIFN call connects to a domestic carrier network, it is directed to a corresponding 8XX number and then terminated domestically. ITFS calls can be
originated from more than 20 international locations, with each country utilizing its own international individual toll free number. With UIFN, customer in the native country use a single toll free number for call origination.
A single toll free number for over 35 countries. That's what Eurus Universal International Free-Phone Number (UIFN) delivers. Reduce costs and market just one toll free number internationally, for greater brand recognition. Our ITFS and UIFN offer both switched and dedicated access. Some of its benefits are:
- A single number from your choice of over 35 countries, with landline, mobile and payphone access
- Customer owned number, allowing for UIFN portability
- Competitive pricing with no monthly fee or call set-up charges
- Flexible technical solutions including ANI, CLID, and OLI
- Available via TDM or VoIP
Out-bound Services by Eurus
VoIP Outbound service directly connects IP networks, enabling our customers to avoid the expense of connecting to TDM equipment. This enables customer to leverage their VoIP network investment and extend their international service reach in a cost effective manner.
The VoIP network is fully integrated with the TDM network; the Carrier VoIP Outbound service benefits from the highly competitive feature set of Eurus's entire suite of services and helps to make the migration to VoIP seamless.
Security is a key feature of Carrier VoIP Outbound service and Eurus has deployed session border controllers (SBCs) to provide a secure interconnection point that protects both Eurus and the customer's networks from each other. High performance is guaranteed because traffic is carried over the Eurus MPLS private backbone in US. This minimizes packet loss, jitter and latency to give the requisite service quality demanded by voice.
- Cisco Gateway
Create Integrated, Scalable Networks Deploy data and voice capabilities in an integrated routing platform to help increase productivity, decrease costs, and lower the cost of ownership.
Cisco Transceiver Modules support Ethernet, Sonet/SDH and Fibre Channel applications across all Cisco switching and routing platforms. Cisco pluggable transceivers offer a convenient and cost effective solution for the adoption in data center, campus, metropolitan area access and ring networks, and storage area networks. Cisco portfolio of hot pluggable interfaces offers a rich set of choices in terms of speeds, protocols, reaches and supported transmission media.The Cisco High Density Digital T1/E1 Voice/Fax Network Module provides a flexible and scalable T1/E1 voice solution for Cisco 2600, 2600XM, 2691, 3600, 3700 and AF5400 etc. series Voice Gateway Routers and supports up to 60 voice channels in a single network module. This network module supports both on-premise and off-premise connections to both Private Branch Exchanges (PBXs) and Public Switched Telephone Networks (PSTNs). For enterprise branches, large businesses and offices wishing to implement robust IP Communications infrastructures, these set of network modules leverage investments in existing legacy telephony equipment and enables the deployment of new packet voice applications while reducing recurring telephony charges.
- Quintum Gateway
Integrates telephony call processing, call control, messaging, contact center and a widely accepted application programming interface into a highly scalable architecture designed to support both circuit-based and IP-based telephony within a distributed enterprise communications network The Quintum UA sends periodic OPTIONS messages to the Primary SIP Server to see if it has “come back.” The Quintum UA will continue to use the Secondary SIP Server until either Primary comes back (responds to OPTIONS msgs w/200-OK) or the Secondary does not respond. There is a possibility of using Quintum hardware for IVR with IXC Billing Center in new version.
Quintum gateways are recommended for all who decided to start VoIP business (calling cards) using analogue lines.
Call to computers behind NAT is also implemented in new version. This function is highly specific and realized
only for PC2phone customers who using IXC dialer. Feature for calls to PC2phone customers is also implemented into the softswitch. Any PC2phone customer can use number for calls redirection if customer is offline.
- PBX Solution
We provide soft PBX solution where we provide services such as Voicemail services with Directory, Call Conferencing, Interactive Voice Response and Call Queuing, Blind transferring, three-way calling, caller ID services, SIP and H.323 through PBX solution.
Softswitch is the concept of separating the network hardware from network software. In traditional circuit switched networks, hardware and software is not independent. Circuit switched networks rely on dedicated facilities for inter-connection and are designed primarily for voice communications. The more efficient packet based networks use the Internet Protocol (IP) to efficiently route voice and data over diverse routes and shared facilities.
Eurus’s VoIP application software empowers wireless and wireline carriers to deliver next-generation voice and multimedia applications and advanced features to increase revenue, enhance competitive differentiation and elevate customer satisfaction. It is a network communication platform that provides a comprehensive range of applications. The Softswitch provide a centralised, scaleable, carrier-grade, call control engine allowing full management and routing of VoIP, SS7 and PSTN traffic in a seamless and integrated environment.
Supporting SIP, H.323, MGCP, MEGACO, SS7 and SIGTRAN enables delivery of secure, reliable VoIP, SS7 and PSTN traffic over multiple disparate networks. Coupled with a robust and powerful signaling engine with leading edge
routing, RADIUS interface, billing, subscriber registration and border control facilities this softswitch delivers a true Class service.
- Intelligent call routing
- Simultaneous media proxy & non-proxy mode
- Protocol manipulation
- Route quality enhancer
- NAT+Firewall traversal
- SIP, SIP-T, H.323, MGCP & MEGACO support
- Integrated SIP proxy & h.323 gatekeeper registration server
- Standard third party billing & routing capabilities
- SNMP support
- Remote management
- High availability & highly scaleable- Full support services
- Online CDR on real time basis
Our online CDR (Call Detail Record) Collection Solution, is a unique, path breaking solution belonging to the mediation suite. CDR Collection and distribution system that utilizes flexibility and adaptability to support
virtually all Wireline and Wireless telecommunications protocols.This solution provides an efficient real time online CDR collection system, which integrates all networking, integrity and security aspects of CDR Collection that are central to various business processes of a Telecom Service Provider.
Eurus solution is based on highly scalable and distributed architecture. The system monitors the SS7 links in the network at various levels on various protocols, and captures the messages to build CDRs. The CDRs can be formatted to comply with any switch CDR format. The data generated by the solution can be used by various applications like
revenue assurance, interconnect billing verification, traffic analysis and SS7 accounting.
SS7 Monitoring Solution provides the operator with an alternate source of data to the Switch generated CDRs, to perform various validations across the network relating to revenue assurance and network performance.
A highly efficient and performance optimized monitor engine ensures that there is absolutely no message loss. It ensures that all the messages are captured, decoded and stored in the database.
Some of the IP networks bandwidth-saving schemes are as follows:
It is a technique used for VoIP which is different from those employed by Frame Relay access devices. Prioritization is directly related to QoS. The key IP QoS protocol was RSVP, which allowed the sender to request a certain set of traffic-handling characteristics for traffic flow, but was not widely adopted. Today, the Eurus Network Services is developing a simpler, more promising solution. The Differentiated Services Model uses the Type of Service (ToS) octet field of the IP header to classify traffic at the borders between the customer and service provider or Internet service providers (ISPs). Currently, there is still no viable QoS for IP services.
- IP Fragmentation
This is performed in a similar fashion as Frame Relay fragmentation. Fragmentation, although required to reduce the overall delay of voice traffic, adds a lot of overhead to IP transmissions due to the large size
of IP headers. IP voice traffic therefore consumes 50% more WAN bandwidth than Frame Relay voice traffic. However, as IP matures, header compression and improved routers will eliminate these shortcomings.
- Available via TDM or VoIP
- Jitter buffer, silence suppression and echo cancellation
These techniques are similar to those employed in VoFR. Echo cancellation is extremely important in VoIP, which often suffers from long network delays.
- Voice compression
It is vital in Voice over IP because traffic usually travels over low-speed links. Some small and medium-sized enterprises, for example, may be connected to the virtual private network (VPN) at only 28.8 kbps. Microsoft Netmeeting, for example, a popular voice application for PCs and laptops, supports ITU G.723.1 voice compression for transmission over dial-up modems. The ITU G.723.1 standard for voice compression over IP ensures toll quality voice.
Benefits of VOIP:
- VoIP can facilitate tasks that may be more difficult to achieve using traditional networks.
- Call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection.
- VoIP phones can integrate with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio
conferencing, managing address books and passing information.
- Establish an immediate local presence in more than 300 markets across the United State and globally.
- Leverage the coverage and economics of the Internet.
- Design and deliver new applications quickly and cost-effectively.
- Streamline call flows.