- EtherSpeak is a ShoreTel certified partner that provides SIP trunking services called SureTrunk, allowing ShoreTel systems to extend into the cloud.
- SureTrunk offers features like centralized billing, auto-failover redundancy between sites, and "BurstaTrunk" which only bills customers for their highest number of concurrent calls in a month rather than a fixed number of trunks.
- These services provide benefits like savings from more efficient trunk usage, convenience of centralized management, and flexibility to connect ShoreTel systems across locations.
SureTrunk is a native SIP trunking solution for ShoreTel customers provided by EtherSpeak that offers cost savings, flexibility, and reliability benefits over traditional TDM trunking. Key features include centralized billing across sites, access to virtual phone numbers anywhere in North America, pay-as-you-go "BurstaTrunk" billing based on maximum concurrent calls, and automatic call rerouting in a disaster recovery configuration. The document provides examples of customers saving 40-70% on their monthly phone bills by switching to SureTrunk.
This document discusses various options for enabling voice services over LTE networks, including adopting existing VoIP solutions from fixed broadband, using dual-radio "simultaneous voice and LTE" devices, circuit-switched fallback which switches between LTE and legacy networks for calls, and voice over LTE via generic access which tunnels legacy call signaling over LTE without leaving the LTE network. It notes subscriber requirements like replicated telephony services, quality, and ubiquity, as well as carrier requirements like efficiency, complexity, and cost. The options are evaluated based on factors like support for services, quality of service, battery life, control by carriers, and infrastructure requirements.
Fibernetics offers a PBX phone system called the Fibernetics Digital PBX that allows businesses to eliminate monthly phone line charges. As a competitive local exchange carrier, Fibernetics operates its own private voice and data networks that are directly connected to the public switched telephone network. The Fibernetics Digital PBX utilizes this network to provide a full-featured phone system with toll-quality voice and high reliability over internet protocol connections while requiring less bandwidth than typical VoIP systems.
Star Telecom is a Canadian SIP trunking provider based in Toronto. They provide SIP trunking and other services designed for contact centers, including 100% uptime guarantees, network monitoring and failover capabilities, and a feature-rich SIP platform. Star Telecom was founded in 2008 by telecom and contact center specialists to better service the Canadian contact center industry. They route over 30 million contact center calls per month.
Avaya VoIP on Cisco Best Practices by PacketBasePacketBase, Inc.
The document provides an overview of Avaya IP communications and best practices for interoperability with Cisco networks. It discusses key considerations for quality of service including recommended delay, jitter and packet loss thresholds. It also provides guidance on general QoS approaches, IP phone deployment, VLAN configuration, QoS settings for Cisco switches, and best practices for WAN connectivity.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
- EtherSpeak is a ShoreTel certified partner that provides SIP trunking services called SureTrunk, allowing ShoreTel systems to extend into the cloud.
- SureTrunk offers features like centralized billing, auto-failover redundancy between sites, and "BurstaTrunk" which only bills customers for their highest number of concurrent calls in a month rather than a fixed number of trunks.
- These services provide benefits like savings from more efficient trunk usage, convenience of centralized management, and flexibility to connect ShoreTel systems across locations.
SureTrunk is a native SIP trunking solution for ShoreTel customers provided by EtherSpeak that offers cost savings, flexibility, and reliability benefits over traditional TDM trunking. Key features include centralized billing across sites, access to virtual phone numbers anywhere in North America, pay-as-you-go "BurstaTrunk" billing based on maximum concurrent calls, and automatic call rerouting in a disaster recovery configuration. The document provides examples of customers saving 40-70% on their monthly phone bills by switching to SureTrunk.
This document discusses various options for enabling voice services over LTE networks, including adopting existing VoIP solutions from fixed broadband, using dual-radio "simultaneous voice and LTE" devices, circuit-switched fallback which switches between LTE and legacy networks for calls, and voice over LTE via generic access which tunnels legacy call signaling over LTE without leaving the LTE network. It notes subscriber requirements like replicated telephony services, quality, and ubiquity, as well as carrier requirements like efficiency, complexity, and cost. The options are evaluated based on factors like support for services, quality of service, battery life, control by carriers, and infrastructure requirements.
Fibernetics offers a PBX phone system called the Fibernetics Digital PBX that allows businesses to eliminate monthly phone line charges. As a competitive local exchange carrier, Fibernetics operates its own private voice and data networks that are directly connected to the public switched telephone network. The Fibernetics Digital PBX utilizes this network to provide a full-featured phone system with toll-quality voice and high reliability over internet protocol connections while requiring less bandwidth than typical VoIP systems.
Star Telecom is a Canadian SIP trunking provider based in Toronto. They provide SIP trunking and other services designed for contact centers, including 100% uptime guarantees, network monitoring and failover capabilities, and a feature-rich SIP platform. Star Telecom was founded in 2008 by telecom and contact center specialists to better service the Canadian contact center industry. They route over 30 million contact center calls per month.
Avaya VoIP on Cisco Best Practices by PacketBasePacketBase, Inc.
The document provides an overview of Avaya IP communications and best practices for interoperability with Cisco networks. It discusses key considerations for quality of service including recommended delay, jitter and packet loss thresholds. It also provides guidance on general QoS approaches, IP phone deployment, VLAN configuration, QoS settings for Cisco switches, and best practices for WAN connectivity.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
Sip Trunking Getting It Right The 1st TimeGraham Francis
This document discusses considerations for migrating from a traditional telephone system to a SIP trunking solution. It provides an overview of key aspects to evaluate such as measuring call patterns to size trunk requirements, selecting an internet service provider and session border controller, ensuring quality of service for voice traffic, choosing phone system integration options, and disaster recovery capabilities. The document emphasizes that thorough planning and testing are needed to get the SIP trunking implementation right from the start.
This document discusses scaling Diameter signaling for LTE networks. It introduces Diameter signaling controllers as a new product category that can provide more efficient, scalable, and reliable Diameter networks. Diameter signaling controllers are needed at the core and edge of networks to handle all Diameter interfaces and applications. They help address pain points like network outages, overload, and interoperability issues. Centralized routing databases can further help scale Diameter signaling controllers by simplifying provisioning and enabling highly scalable routing of hundreds of millions of entries.
HomePNA is an ITU standard for home networking that allows triple play services over existing phone and coax wires. It has shipped over 10 million chipsets worldwide and supports speeds up to 320Mbps. HomePNA provides quality of service for high definition streaming and remote management capabilities. The HomePNA Special Interest Group includes over 40 companies working to standardize the technology and provide support for service providers implementing HomePNA home networks.
Next Generation Networks (NGNs) connect voice and data communications over a common IP-based network. This allows phone calls to be made over internet connections using Voice-over-IP (VoIP) instead of traditional analog or ISDN connections. Migrating to an NGN requires replacing analog and ISDN phones with IP phones and connecting to the network with a single connection instead of separate lines. Companies must plan their migration to an NGN carefully by assessing their current infrastructure and creating a migration master plan.
This article introduces a view of a generic Service Provider IP distribution system including DVB's IP standard; a comparison of Internet and managed SP IP video distribution; how a broadcaster can inject TV programming into the Internet and, finally, how to control the Quality of Experience of video in an IP network.
The document proposes implementing Active Optical Network (AON) Fiber-To-The-X (FTTx) networks for VNPT to offer triple-play services, increase revenue, and compete effectively. It recommends broadband service tiers for business and home subscribers with throughput from 2Mbps to over 20Mbps, powered by VOLKTEK's market-proven FTTx switches, routers, and gateways. Ring topologies should serve high-ARPU business subscribers, while stars suffice for lower-ARPU homes to reduce costs.
The Polycom PathNavigator is an advanced gatekeeper solution that makes IP and ISDN video conferencing easy to use. It provides benefits such as simplified dialing, automated deployment and management, and intelligent network routing. Key features include conference on demand, least cost routing, call forwarding, and alternate routing to increase flexibility and reduce costs. The solution is designed to integrate with other network components and manage complex video networks efficiently.
The document discusses various voice deployment features in Microsoft Lync Server 2010 including call park, unassigned number routing, E911 location services, private lines, caller ID controls, phone infrastructure requirements, voice routing considerations, and mediation server consolidation. Planning items covered include branch resiliency, datacenter resiliency, call admission control, topology changes, enhanced 911 for North America, analog devices, common area phones, and malicious call trace.
Inbound services allow businesses to manage incoming calls even during disruptions by routing calls to alternative phone numbers. Key features include call forwarding, voicemail delivery by email, call recording, and interactive voice response systems. Businesses can set up geographic or non-geographic inbound numbers and build call routing plans using options like call queues, hunt groups, and automated attendants. Pricing is flexible with options to record some calls or build a custom solution. Inbound aims to improve customer service, make businesses more resilient, and provide effective call management.
Verizon has strict requirements for IP/MPLS-based carrier Ethernet networks providing public Ethernet services. These include supporting large numbers of customers and interfaces through scalable architectures like PBB and MPLS. Services must conform to MEF definitions and provide high reliability through fast convergence and OAM. Extensive testing and certification are required to ensure performance meets standards.
The Mobile Network’s Founder and Editor, Keith Dyer, joins Syniverse’s Chief Marketing Officer, Mary Clark, and Senior Solutions Engineer, Leo Casey, this week to help mobile operators better understand the future of roaming and charging settlement for VoLTE.
SIP Trunking & Security in an Enterprise NetworkDan York
How secure are your VoIP systems as you deploy SIP-based systems in an enterprise environment? In this slide deck presented by VOIPSA Best Practices Chair Dan York at the Ingate SIP Trunking Seminars at ITEXPO September 17, 2008, Dan York walks through the security issues related to VoIP (with a focus on SIP trunking), the tools out there to attack/test VoIP systems, best practices and resources. (An audio recording of this session was made and will be available.)
Introduction to Diameter: The Evolution of SignalingPT
As telecommunications networks have advanced, so have the signaling procedures. This introduction to Diameter gives you an overview of the evolution of signaling.
Minimizing Server Throughput for Low-Delay Live Streaming in Content Delivery...Gwendal Simon
Large-scale live streaming systems can experience bottle- necks within the infrastructure of the underlying Content Delivery Network. In particular, the “equipment bottleneck” occurs when the fan-out of a machine does not enable the concurrent transmission of a stream to multiple other equipments. In this paper, we aim to deliver a live stream to a set of destination nodes with minimum throughput at the source and limited increase of the streaming delay. We leverage on rateless codes and cooperation among destination nodes. With rateless codes, a node is able to decode a video block of k information symbols after receiving slightly more than k encoded symbols. To deliver the encoded symbols, we use multiple trees where inner nodes forward all received symbols. Our goal is to build a diffusion forest that minimizes the transmission rate at the source while guaranteeing on-time delivery and reliability at the nodes. When the network is assumed to be lossless and the constraint on delivery delay is relaxed, we give an algorithm that computes a diffusion forest resulting in the minimum source transmission rate. We also propose an effective heuristic algorithm for the general case where packet loss occurs and the delivery delay is bounded. Simulation results for realistic settings show that with our solution the source requires only slightly more than the video bit rate to reliably feed all nodes.
The document discusses preparing a WLAN infrastructure to support voice services. It outlines several key steps:
1) Define voice service requirements such as number of users, areas of coverage, and expected traffic volumes.
2) Ensure the wired network is ready to support voice with features like QoS, PoE, and separate voice VLANs.
3) Design the WLAN for dense, pervasive coverage using tools to accurately model signal propagation. A dual overlay network with separate SSIDs for voice and data is recommended.
VOIP Presentation Tjcomms.Co.Uk Hosted TelephonyTyler James
Hosted Telephone Systems or VoIP will bring all your communications together across all your locations including home-workers ensuring that you now have one single phone system for your business, which all calls between users are free irrespective of their global destination.
With a traditional telephone system, the day it goes on the wall is the day it went out of date! With Hosted Telephone Systems you connect to it, so you are always have the latest communication offerings at your disposal within your business. As and when we upgrade, you get upgraded.
The document discusses enterprise E911 capabilities and solutions. It explains that IP telephony systems now allow for more complex environments with multiple buildings and locations, requiring E911 applications to properly route 911 calls and provide accurate location information to dispatch centers. These applications allow grouping of users to share location records and emergency response locations. They also provide benefits like automatic MAC address management and real-time 911 call notifications. The document reviews E911 capabilities and tools available in various IP PBX platforms like Cisco, Avaya, Nortel, and Mitel.
PAETEC Disaster Recovery & Business Continuity SolutionsMark Lawrence Peay
The document discusses PAETEC's disaster recovery and network diversity solutions. It outlines critical BC/DR elements and assessing business risks. It then describes PAETEC's products that provide inherent and customized network diversity, including access diversity solutions, call rerouting solutions, MPLS/internet redundancy, and email scanning backup. Sample customer architectures are provided applying these solutions.
The document discusses SIP trunking and issues related to its implementation. It describes SIP trunking technology, issues around quality of service, survivability, network management, integration with legacy PBXs, and IP PBX interoperability. Solutions discussed include the SmartNode DownStreamQoS, auto-provisioning, and the SmartNode SN4960 gateway and SN5200 session border controller to address these issues and provide SIP trunking capabilities to different PBX environments.
The document provides an overview and training on selling VoIP services commercially. It discusses what VoIP is, why customers would use the service to save money on calling features. It outlines the sales process, including qualifying questions to ask customers. It also summarizes the key product offerings of hosted PBX, SIP trunking, features and pricing.
Sip Trunking Getting It Right The 1st TimeGraham Francis
This document discusses considerations for migrating from a traditional telephone system to a SIP trunking solution. It provides an overview of key aspects to evaluate such as measuring call patterns to size trunk requirements, selecting an internet service provider and session border controller, ensuring quality of service for voice traffic, choosing phone system integration options, and disaster recovery capabilities. The document emphasizes that thorough planning and testing are needed to get the SIP trunking implementation right from the start.
This document discusses scaling Diameter signaling for LTE networks. It introduces Diameter signaling controllers as a new product category that can provide more efficient, scalable, and reliable Diameter networks. Diameter signaling controllers are needed at the core and edge of networks to handle all Diameter interfaces and applications. They help address pain points like network outages, overload, and interoperability issues. Centralized routing databases can further help scale Diameter signaling controllers by simplifying provisioning and enabling highly scalable routing of hundreds of millions of entries.
HomePNA is an ITU standard for home networking that allows triple play services over existing phone and coax wires. It has shipped over 10 million chipsets worldwide and supports speeds up to 320Mbps. HomePNA provides quality of service for high definition streaming and remote management capabilities. The HomePNA Special Interest Group includes over 40 companies working to standardize the technology and provide support for service providers implementing HomePNA home networks.
Next Generation Networks (NGNs) connect voice and data communications over a common IP-based network. This allows phone calls to be made over internet connections using Voice-over-IP (VoIP) instead of traditional analog or ISDN connections. Migrating to an NGN requires replacing analog and ISDN phones with IP phones and connecting to the network with a single connection instead of separate lines. Companies must plan their migration to an NGN carefully by assessing their current infrastructure and creating a migration master plan.
This article introduces a view of a generic Service Provider IP distribution system including DVB's IP standard; a comparison of Internet and managed SP IP video distribution; how a broadcaster can inject TV programming into the Internet and, finally, how to control the Quality of Experience of video in an IP network.
The document proposes implementing Active Optical Network (AON) Fiber-To-The-X (FTTx) networks for VNPT to offer triple-play services, increase revenue, and compete effectively. It recommends broadband service tiers for business and home subscribers with throughput from 2Mbps to over 20Mbps, powered by VOLKTEK's market-proven FTTx switches, routers, and gateways. Ring topologies should serve high-ARPU business subscribers, while stars suffice for lower-ARPU homes to reduce costs.
The Polycom PathNavigator is an advanced gatekeeper solution that makes IP and ISDN video conferencing easy to use. It provides benefits such as simplified dialing, automated deployment and management, and intelligent network routing. Key features include conference on demand, least cost routing, call forwarding, and alternate routing to increase flexibility and reduce costs. The solution is designed to integrate with other network components and manage complex video networks efficiently.
The document discusses various voice deployment features in Microsoft Lync Server 2010 including call park, unassigned number routing, E911 location services, private lines, caller ID controls, phone infrastructure requirements, voice routing considerations, and mediation server consolidation. Planning items covered include branch resiliency, datacenter resiliency, call admission control, topology changes, enhanced 911 for North America, analog devices, common area phones, and malicious call trace.
Inbound services allow businesses to manage incoming calls even during disruptions by routing calls to alternative phone numbers. Key features include call forwarding, voicemail delivery by email, call recording, and interactive voice response systems. Businesses can set up geographic or non-geographic inbound numbers and build call routing plans using options like call queues, hunt groups, and automated attendants. Pricing is flexible with options to record some calls or build a custom solution. Inbound aims to improve customer service, make businesses more resilient, and provide effective call management.
Verizon has strict requirements for IP/MPLS-based carrier Ethernet networks providing public Ethernet services. These include supporting large numbers of customers and interfaces through scalable architectures like PBB and MPLS. Services must conform to MEF definitions and provide high reliability through fast convergence and OAM. Extensive testing and certification are required to ensure performance meets standards.
The Mobile Network’s Founder and Editor, Keith Dyer, joins Syniverse’s Chief Marketing Officer, Mary Clark, and Senior Solutions Engineer, Leo Casey, this week to help mobile operators better understand the future of roaming and charging settlement for VoLTE.
SIP Trunking & Security in an Enterprise NetworkDan York
How secure are your VoIP systems as you deploy SIP-based systems in an enterprise environment? In this slide deck presented by VOIPSA Best Practices Chair Dan York at the Ingate SIP Trunking Seminars at ITEXPO September 17, 2008, Dan York walks through the security issues related to VoIP (with a focus on SIP trunking), the tools out there to attack/test VoIP systems, best practices and resources. (An audio recording of this session was made and will be available.)
Introduction to Diameter: The Evolution of SignalingPT
As telecommunications networks have advanced, so have the signaling procedures. This introduction to Diameter gives you an overview of the evolution of signaling.
Minimizing Server Throughput for Low-Delay Live Streaming in Content Delivery...Gwendal Simon
Large-scale live streaming systems can experience bottle- necks within the infrastructure of the underlying Content Delivery Network. In particular, the “equipment bottleneck” occurs when the fan-out of a machine does not enable the concurrent transmission of a stream to multiple other equipments. In this paper, we aim to deliver a live stream to a set of destination nodes with minimum throughput at the source and limited increase of the streaming delay. We leverage on rateless codes and cooperation among destination nodes. With rateless codes, a node is able to decode a video block of k information symbols after receiving slightly more than k encoded symbols. To deliver the encoded symbols, we use multiple trees where inner nodes forward all received symbols. Our goal is to build a diffusion forest that minimizes the transmission rate at the source while guaranteeing on-time delivery and reliability at the nodes. When the network is assumed to be lossless and the constraint on delivery delay is relaxed, we give an algorithm that computes a diffusion forest resulting in the minimum source transmission rate. We also propose an effective heuristic algorithm for the general case where packet loss occurs and the delivery delay is bounded. Simulation results for realistic settings show that with our solution the source requires only slightly more than the video bit rate to reliably feed all nodes.
The document discusses preparing a WLAN infrastructure to support voice services. It outlines several key steps:
1) Define voice service requirements such as number of users, areas of coverage, and expected traffic volumes.
2) Ensure the wired network is ready to support voice with features like QoS, PoE, and separate voice VLANs.
3) Design the WLAN for dense, pervasive coverage using tools to accurately model signal propagation. A dual overlay network with separate SSIDs for voice and data is recommended.
VOIP Presentation Tjcomms.Co.Uk Hosted TelephonyTyler James
Hosted Telephone Systems or VoIP will bring all your communications together across all your locations including home-workers ensuring that you now have one single phone system for your business, which all calls between users are free irrespective of their global destination.
With a traditional telephone system, the day it goes on the wall is the day it went out of date! With Hosted Telephone Systems you connect to it, so you are always have the latest communication offerings at your disposal within your business. As and when we upgrade, you get upgraded.
The document discusses enterprise E911 capabilities and solutions. It explains that IP telephony systems now allow for more complex environments with multiple buildings and locations, requiring E911 applications to properly route 911 calls and provide accurate location information to dispatch centers. These applications allow grouping of users to share location records and emergency response locations. They also provide benefits like automatic MAC address management and real-time 911 call notifications. The document reviews E911 capabilities and tools available in various IP PBX platforms like Cisco, Avaya, Nortel, and Mitel.
PAETEC Disaster Recovery & Business Continuity SolutionsMark Lawrence Peay
The document discusses PAETEC's disaster recovery and network diversity solutions. It outlines critical BC/DR elements and assessing business risks. It then describes PAETEC's products that provide inherent and customized network diversity, including access diversity solutions, call rerouting solutions, MPLS/internet redundancy, and email scanning backup. Sample customer architectures are provided applying these solutions.
The document discusses SIP trunking and issues related to its implementation. It describes SIP trunking technology, issues around quality of service, survivability, network management, integration with legacy PBXs, and IP PBX interoperability. Solutions discussed include the SmartNode DownStreamQoS, auto-provisioning, and the SmartNode SN4960 gateway and SN5200 session border controller to address these issues and provide SIP trunking capabilities to different PBX environments.
The document provides an overview and training on selling VoIP services commercially. It discusses what VoIP is, why customers would use the service to save money on calling features. It outlines the sales process, including qualifying questions to ask customers. It also summarizes the key product offerings of hosted PBX, SIP trunking, features and pricing.
The document summarizes an IP phone system called Allworx that is designed for businesses. It provides diverse voice services, robust features, and global integration capabilities across multiple sites. Allworx systems aim to satisfy customer needs with low total cost of ownership through easy installation and maintenance.
This document provides an overview of the Allworx phone system. Some key points:
- Allworx was founded in 1998 and is now owned by Windstream. They focus on phone systems for small to medium businesses.
- Their systems include the Allworx 6x, 6x12, and 48x servers which support up to 12, 60, and 250 users respectively. Phones include the 9224, 9212L, 9204, and 9202E models.
- Features include SIP trunking, unified messaging, conferencing, mobility applications, call routing, presence settings and more. Advanced features include multi-site functionality across 99 servers, automatic call distribution, and conference center options.
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
VOIP allows IP networks to carry voice applications like telephony and conferencing. It uses protocols like SIP, H.323, and MGCP for signaling and codecs like G.711 and G.729 for compressing analog voice. Key VOIP components include IP phones, gateways, call agents, and MCUs. Signaling protocols establish and terminate sessions, with SIP and H.323 using a peer-to-peer model and MGCP using a client-server model. Considerations for VOIP include low jitter, latency under 150ms, minimal packet loss, and high availability to provide a reliable voice service over IP networks.
Comcast Business Class Trunks Pri Customer Presentation 0911 (2)Henry Garcia
- Comcast offers business class trunks that allow businesses to converge their voice and data services over Comcast's network.
- The trunks provide a switched digital local voice trunk service with an ISDN Primary Rate Interface.
- Benefits include cost savings from converged services, scalability, reliability, and future proofing a business's communications.
Comcast Business Class Trunks PRI PresentationMiguel Spencer
Comcast Business Class Trunks are a switched voice trunk service with ISDN/PRI connectivity from a customer’s Private Branch Exchange (PBX) to the Comcast network, providing a flexible and intelligent way to maximize next generation voice services based on Comcast’s advanced IP network.
Why Comcast Business Class PRI Trunks?
Network Reliability: Comcast’s privately owned and manage network lets us proactively monitor all elements as we strive for maximum PRI service availability
Business Voice Continuity: All voice networks are geographically redundant and phone calls can be routed between networks on a call-by-call basis providing seamless service during network impairments
Service Level Agreement: Your Service Level Agreement will define measurable service assurance and service restoral policies to define faster response times
Security: Voice services are prioritized over all other traffic and all network equipment is secured by 2 layers of authentication
Scalability: Buy what you need when you need it – we let you buy in single channel increments, others do not
Cost Effective: Save money by bundling with other Comcast services – Data or Internet
Easy Integration: Use your existing PBX to consolidate business lines to a single facility for integrated voice and data capabilities
Email: miguel_spencer@cable.comcast.com or Call +1 786 512 6913
VoIP allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting the voice signal from analog to digital, breaking it into packets, sending it over IP, reassembling it at the destination, and converting it back to analog. VoIP has advantages like low cost and portability but disadvantages like quality issues during power outages or network instability. Major challenges include addressing latency, echo, jitter, connection problems through firewalls and NAT, and overall reliability.
Mohammad Faisal Kairm(073714556) Assignment 2mashiur
This document discusses implementing IP telephony over traditional PSTN and PLMN networks to reduce transmission costs. It proposes replacing TDM connections between exchanges with an IP backbone using voice compression to utilize the full bandwidth of E1 circuits. Some challenges of implementing an IP backbone for voice include ensuring sufficient network quality by limiting packet loss, jitter, and latency to avoid voice quality issues like dropouts and echo. Overall, IP telephony could lower costs through more efficient use of existing infrastructure and management capabilities.
This document discusses the drivers behind converging voice and data networking. It describes drawbacks of the public switched telephone network (PSTN) including its inability to quickly deploy new features or converge data, voice, and video. The document outlines how IP networking provides a more flexible architecture and open standards to enable this convergence compared to the proprietary nature of the PSTN. Key components that enable voice over IP like RTP, call control protocols, and an open application layer are also summarized.
What is SIP Trunking?
How good is the ROI?
Benefit #1: Local Phone Numbers with Centralized Call Management
Benefit #2: Control and Security
Benefit #3: Increased productivity and collaboration
Benefit #4: Scalability
Benefit #5: Faster Disaster Recovery
Benefit #6: Foundation for Unified Communications / WebRTC
WebRTC Takes UC Further – With Ingate the Future is Soon
This document summarizes Sunesh Kumra's thesis presentation on load balancing in IP protocols. The presentation covered:
1) Introduction to load balancing concepts like stateless vs stateful load balancing, and types of load balancers.
2) The research problem of balancing different types of traffic like UDP, long TCP sessions, and short TCP sessions across server nodes. Requirements like ensuring a single session goes to one node.
3) Approaches for stateless load balancing including network address translation, IP tunneling, and direct routing. Stateful load balancing requires maintaining call state information.
4) Challenges of dynamically adding or removing nodes, where the hash function could route an existing session to a
This document summarizes Sunesh Kumra's thesis on load balancing in IP protocols. It discusses different types of load balancers including stateless and stateful approaches. It covers dynamic addition and removal of nodes, capacity-based balancing, and overload control. The conclusion emphasizes that load balancers need congestion control and scalability to add nodes without interrupting traffic as IP telephony systems become more distributed.
This document provides an outline for a lecture on Transmission Control Protocol (TCP). It discusses TCP's role in providing reliable, in-order delivery of data between applications on different hosts. Key topics covered include TCP segments, ports, sockets, flow control using sliding windows, congestion control, connection establishment and termination procedures. Diagrams illustrate TCP state transitions and the format of TCP packet headers.
Key factors to be considered by communication service providers while implementing switching & routing infrastructure for enterprise global voice solutions
This document discusses Xcel Energy's implementation of a new IVR system to better handle peak call volumes. Some key points:
- The new system can handle up to 1700 calls simultaneously using Avaya technology, compared to the previous outsourced solution that could only handle 200,000 calls.
- It provides self-service options like outage logging and payment information. During high volumes, calls will be routed to the new in-house system or outsourced provider for assistance.
- The system uses automatic triggers and overflow routing to outbound lines to maintain service levels during peaks. It reduced costs by bringing functions in-house and improved the customer experience.
IP telephony has received interest from many users and organizations as it provides cost savings over traditional phone lines. VoIP saves money by using existing computer networks and IP infrastructure rather than separate phone lines, reducing line charges, feature charges, taxes, and fees. Many organizations currently maintain separate networks for data and voice, but integrating the two using VoIP provides a more cost effective and flexible unified solution.
1. InTechnology is a telecommunications service provider that offers calls and lines, IP telephony, data centers, and managed network services with a focus on immediate savings, enhanced customer service tools, migration from expensive ISDN lines to cheaper SIP trunks, and future-proofing through cloud-based services.
2. Their calls and lines proposition aims to deliver immediate call cost savings of up to 40% through tariff analysis, enhanced customer service tools through their INBOUND portal, savings of up to 70% by migrating from ISDN to SIP trunks, and future-proofing by upgrading existing on-premise systems to their cloud-based offerings.
3. They provide a full portfolio of IT and
1. SureTrunk for ShoreTel
Introducing an Easy Path to
911 Compliance with SIP
Neil Darling
EtherSpeak
(866) Ether-IP
shoretel@ietherspeak.com / ndarling@ietherspeak.com
2. Who Are We?
• EtherSpeak is Your ShoreTel Certified
Technology Partner for Enabling Cloud Based
Services Including Voice, Video Bridging and
Fax-Over-IP
• Only Certified ShoreTel Technology Provider that
is ShoreTel “Customer Proven” & “Partner
Proven”
• We provide SIP Trunks – or “Business-
Telephone-Lines-Over-Internet” allowing for
enterprise savings consolidation or elimination of
legacy POTS / PRI; Advanced Redundancy; and
International DID routing options for customers
3. Solving the 911 Problem…
• Customers using VoIP on large enterprise networks are having
difficulty maintaining VoIP savings with distributed POTS lines
dedicated strictly for 911 compliance
• Virtual Numbering Option: Utilize 911 compliant virtual local
numbers, available from virtually any area code in the USA / CAN
• Centralize costs but distribute benefits – providing 911 compliance
for a fraction of the cost of hundreds of POTs lines
• EtherSpeak centralizes the billing – even for hundreds of offices
on one easy to read bill
• BurstaTrunk Option: Customers only billed monthly for amount
“high-water mark” of concurrent calls – eliminate waste and show
tangible ROI that is key for technology refresh and closing the deal
4. What Value Does EtherSpeak
Bring to ShoreTel Customers?
• Cost Savings: SureTrunk provides no CapEx and
administrative savings with SureTrunk Centralized Billing
• Redundancy: SureTrunk provides seamless re-direction in
the event of an outage, or on-site problem - connects to
another site upon error condition (RedundaTrunk service
option)
• Affordability: SureTrunk provides High-Water Mark Billing.
Customers only pay at highpoint of what they use
concurrently (BurstaTrunk service option)
• Flexibility: SureTrunk provides seamless connectivity for
ShoreTel; Providing 911 compliant DID access to most of
North America
5. Real Life Example of Value of SureTrunk
with BurstaTrunk for 911 Compliance. . .
• EtherSpeak SureTrunk 911 w/
• Current Carrier BurstaTrunk
– 27 offices – each with two – Customer maintains 6 trunks
POTs lines for 911 ~$50.00 and enables Burstatrunk
each or ~$1500 / month Option
– Lines are not integrated to – DID’s delivered for each office
ShoreTel – They are not even that route to local PSAP or
used regularly – just a sunk 911 call center
cost – Installed in 90 minutes
– Customer is over-provisioned – Numbers are $2.00 each or
$54.00 monthly cost across
the enteprise
– - Customer only provisioned
for 6 trunks – with availability
for hundreds at $146.00 /
month
This customer is saving approx. $1,200.00 month or $14,400/
year! What impact does this have on your budget if you have an
extra $14,400?!?
6. ShoreTel and BurstaTrunk
ShoreTel Client Office A Legacy
PSTN
POTS / PRI
IPSEC Tunnel Customer provided with ability to make
for Signaling any number of simultaneous calls –
Carrier MPLS
SureTrunk Connection - providing they have ShoreTel SIP Hand-off
licenses and adequate bandwidth for
highest amount of expected calls.
EtherSpeak Core
ShoreGear Internet / VPN Router
Main Office – MPLS
ShoreTel Director &
- 6 Trunks with a “Burst” - Tunnel.a
192.168.0.0/24 Office Subnet
Calls calculated on 6 second billing
increment
Inbound / Local Free; Customer
only billed on 1 + dialing
& inbound toll-free utilization
EtherSpeak
BurstaTrunk Billing Results
Session
Border
Controller
All Sites on One Bill!
PSTN
Gateways
VoIP PSTN
Peering Origination
and
Termination
- Customer only billed for 33 concurrent calls for given month!
- Compared with a 46 Channels / PRI, Customer would save on SureTrunk Monthly Concurrent Call
Calculation Based on Customer CDRs
14 unused trunks in this monthly example
7. ShoreTel and e-Billing
ShoreTel Client Office A Legacy
PSTN
POTS / PRI Customer provided with one bill
IPSEC Tunnel regardless of number of sites
for Signaling
SureTrunk with BurstaTrunk option Carrier MPLS
SureTrunk Connection
provides ShoreTel customers the most Hand-off
cost-effective method for managing
monthly recurring expense
EtherSpeak Core
ShoreGear Internet / VPN Router
Main Office – MPLS
ShoreTel Director &
- 6 Trunks with a “Burst” - Tunnel.a
192.168.0.0/24 Office Subnet
Calls calculated on 6 second billing
increment
Inbound / Local Free; Customer
only billed on 1 + dialing
& inbound toll-free utilization
SureTrunk Centralized Billing
EtherSpeak
Session
Border
Controller
All Sites on One Bill!
PSTN
Gateways
VoIP PSTN
Peering Origination
and
- Convenience of one bill to pay for all customer offices Termination
- Enables with BurstaTrunk, customer is able to only pay for SureTrunk Monthly provided on one easy to undertsand bill
trunks used – no more wasted capacity and easy to use customer portal
8. ShoreTel and Virtual 911 Numbering
ShoreTel Client Office A Legacy
PSTN
POTS / PRI
IPSEC Tunnel Customer provided with DIDs from
for Signaling virtually any LATA in USA / CAN / UK
Carrier MPLS
SureTrunk Connection - All calls flow inbound (originate) from Hand-off
EtherSpeak to customer ShoreTel
over Internet or MPLS connection
EtherSpeak Core
VPN Router
ShoreGear Internet / - DIDs have option for
MPLS inbound only – or 911
ShoreTel Director with DID numbers compliance with auto
availability for 95% USA and 80% of CAN Tunnel.a routing to local 911 center
(PSAP).
192.168.0.0/24 Office Subnet
- Inbound only: numbers
are for inbound routing
only and should not show
up in caller-id
SureTrunk VN Features:
Inbound / Local Free; Customer
EtherSpeak only billed on 1 + dialing
& inbound toll-free utilization
Session
Border
Controller
All Sites on One Bill – or
One Bill per Site
PSTN
Gateways
VoIP PSTN
Peering Origination
and
Termination
SureTrunk Monthly Concurrent Call
- EtherSpeak provides direct-dial number availability in Calculation Based on Customer CDRs
95% of USA, 80% of CAN and 50% of UK
9. SureTrunk Disaster Recovery
ShoreTel Client Office A Legacy
PSTN
POTS / PRI VPN tunnel terminated from primary
IPSEC Tunnel and secondary customer locations to
EtherSpeak’s edge.
Carrier MPLS
In the event of connectivity or site Hand-off
Primary ShoreTel Switch to failure – inbound calls routed to
SureTrunk Connection secondary SIP switch; or tertiary PRI /
POTS through PSTN
EtherSpeak Core
ShoreGear Internet / VPN Router
Main Office – WAN MPLS
ShoreTel Director & SureTrunk DR
Main SIP Switch location Tunnel.a provides 2 call
paths, a local DID;
192.168.0.0/24 Office Subnet Tunnel.b free-local; free-
inbound and 100
minutes of included
ShoreTel Client Office B 1 + long distance
IPSEC Tunnel With DR enabled,
Encryption Domain ShoreTel
Src: 192.168.10.b Dst: 172.26.a.b System auto- customers recover
detects error from PSTN
EtherSpeak condition outages quickly
Session and painlessly
Secondary ShoreTel Switch to
SureTrunk Connection
Border
Controller
ShoreGear
Remote Office 74.84.203.x/24
Backup SIP Switch location Legacy
PSTN PSTN
192.168.10.0/24 Office Subnet POTS / PRI Gateways
PSTN
VoIP Origination
If primary connection fails – EtherSpeak will send and receive calls from site B’s connection and
Peering and
secondary ShoreTel switch
Termination
Outbound calls routed to secondary ShoreTel switch in the event of failure
Inbound calls automatically directed to secondary ShoreTel switch / tunnel – or via PSTN to POTS / PRI
10. Next Steps:
- No Obligation Trial Provided –
Just send an email to:
ShoreTel@ietherspeak.com to
request free trial
11. Neil Darling
www.suretrunk.com
• 703-649-0025 Direct Dial / 703-795-6111 Cell
• ndarling@ietherspeak.com
www.suretrunk.com www.switchtrunk.com www.ziptrunk.com www.ocstrunk.com