1 1 High Availability, Security and Peak Performance in Hosted VoIP Deployments Introduction Technical White Paper The hosted VoIP market continues to grow dramatically with more and more providers offering enterprise-like IP PBX functionality as an easy-to-deploy, affordable monthly service. The appeal of such solutions to small and mid-size businesses are many; but primarily, by outsourcing all of the management aspects of IP-based telephony, businesses can save costs and eliminate the worry and headaches associated with voice infrastructure management. At the same time, these VoIP providers must meet the challenges of delivering reliable services and exceptional customer support. These challenges include: Maximizing IP phone service availability and SIP Survivability Preventing -related security attacks Maintaining peak VoIP performance and voice quality while using common links Managing voice-data convergence issues when a new or prospective VoIP customer has an existing data network installed Providing built-in serviceability so that customer problems can be resolved quickly The main elements of hosted VoIP service delivery are outlined below. This paper will identify the various problems and challenges that service providers face today and how the 365-VOIP from Aspen Networks can be utilized to deliver high availability, security, and peak performance. Elements of Service Delivery Typically, a hosted VoIP service provider installs IP phones connected to an Ethernet at the business customer s premises. A data for PCs connected to the usually exists as well. A WAN or broadband router connects the to the and at least one last mile link terminates in this router. The IP phones communicate either with a hosted IP PBX (SIP server such as Broadsoft) or SIP-based Session Border Controller (examples are Acme Packet or Juniper/Kagoor) located at the hosted VoIP service provider s data center. These elements of hosted IP PBX service, and how they connect, are depicted in Figure 1 below.
2 2 Data Center of VoIP Provider Hosted IP PBX (SIP Server or Session Border Controller) Elements in the Delivery of Hosted IP PBX Service Customer Premises IP Phones Fig 1 Typical Elements in Hosted IP PBX Service Challenges from a Provider Perspective The use of the as depicted above presents several challenges: 1. There are several points of failure in the path between the IP phone and its SIP server. 2. It is possible for the IP phones to come under a denial of service or other security attack from the if precautions are not taken. 3. If the data becomes compromised by malware, infected PCs could be used as a backdoor to launch attacks into the service provider s IP PBX infrastructure. 4. Mixed voice and data traffic can present convergence challenges. Large bursts of data or video traffic can adversely affect VoIP if allowed on the same link. To meet these challenges VoIP providers must address all aspects of reliability, availability and serviceability. What follows is a detailed discussion of how redundancy, security, WAN traffic optimization and customer premise management can all contribute to delivering a superior hosted VoIP service.
3 1. VoIP Reliability and Maximizing Service Uptime Fig 2 below identifies 7 different points of failure in a hosted VoIP deployment: 3 1. The customer premises WAN/broadband router it can simply fail, or come under various Denial of Service attacks that cause it to hang until it is power-cycled or rebooted. 2. The last mile link this could be a T1 line, DSL, cable provider, or wireless link. These links can fail outright or degrade to the extent that voice quality breaks. 3. The customer s local ISP can contribute to serious failures, the most common of which are ISP-owned router failures, with outages sometimes lasting for hours. There are also transient periods of congestion and packet loss when the local ISP s links to the backbone are under-provisioned or temporarily oversubscribed. 4. The issue of local ISP failure (see 3 above) also applies to the ISP that provides connectivity at the data center used by the hosted IP PBX provider. 5. Last mile link failure (see 2 above) can also occur at the IP PBX/Data Center end. 6. The issue of first-hop router failure (see 1 above) could also apply at the IP PBX/Data Center end. 7. Lastly, there is the IP PBX at the Data Center itself. It may fail, or its software may crash and hang. Or it may be temporarily taken down for a scheduled maintenance. Point(7) Data Center of VoIP Provider SIP Server or Session Border Controller Point(6) Point(5) Point(4) SIP Reliability and Survivability Challenges Customer Premises IP Phones Point(1) Point(2) Point(3) Fig 2 The 7 Different Points in Service Delivery
4 4 The enables service providers to meet all of these challenges through cleverly managed redundancy of links, routers and servers coupled with auto-recovery and intelligent voice traffic management. By deploying the Application Layer Gateway as a transparent SIP proxy at the customer premises problems are contained and managed at a single point. The solution is illustrated in Fig 3 below. Data Center 1 Main SIP Server or Session Border Controller Data Center 1 Data Center 2 Main SIP Server or Session Border Controller Main SIP Server or Session Border Controller Customer Premises Customer Premises Fig 3 SIP Application Layer Gateway Solution SIP ALG Several features are worth highlighting: 1. Last mile link, router and ISP failures are auto-recovered transparently at the application layer by the, as long as a redundant path is available. 2. No complex router solutions (such as BGP) are needed, allowing use of broadband (DSL, cable, wireless) mixed with T1 links, or all-broadband (up to 4 DSL or Cable links can be load balanced by the 365-VOIP) 3. Data Center and SIP server Survivability is also handled failures of a hosted IP PBX or Session Border Controller are detected and corrected in less than 6 seconds, in a manner transparent to the IP phones. As an example, it may become necessary to urgently update the SIP server software, or its operating system, to improve network security or avoid attacks. This requires a scheduled shut down, which is disruptive to users. With Aspen s SIP survivability feature, IP phones will always get dial tone the fail over is performed immediately and transparently. 4. Instability Detection: Not merely ISP up/down events, but various instabilities such as packet loss or flip-flop behavior on the link are also handled by automatically moving the RTP media streams from the IP phones to an alternate, more stable link 5. Flexible Data Center Configurations are supported IP phones can be configured to connect to the SIP server directly or indirectly via a Session Border Controller. The NAT Traversal and SIP Header fix-up can be optionally turned
5 5 ON or OFF. Where the service provider chooses not to deploy Session Border Controllers, the Aspen SIP ALG needs to be fully enabled for NAT Traversal. 6. /Data Convergence: The Data and VoIP traffic can be configured to go on separate links if needed; it is possible to give priority to voice over data in cases where voice and data mix. It is also possible to allow data to fail over to the voicepreferred link, or to disallow such fail over. 7. Registration Rate Pacing: The Aspen SIP ALG acts as a transparent SIP registration proxy for all the IP phones at the customer premises, and will perform SIP registration rate pacing to throttle down the volume of traffic hitting the SIP server at the data center. 8. Graceful Fail Back: When the policy-preferred link for VoIP fails, IP phone traffic is moved to the next link in order of preference. When later, the preferred link is restored and operational again, RTP media streams are moved back to the preferred link in a graceful manner (care is taken to not break existing VoIP calls). 9. Application Consistency during SIP Fail Over: All key applications used in hosted IP Telephony Call Transfer, Conference Calls, Shared Call Appearances, Mail to name a few survive automatically when either an link or SIP server fails or is brought down for maintenance. 10. Transparent SIP Proxy: By using a transparent SIP proxy architecture, a much more robust system of IP phone configuration management is empowered. The Aspen 365 auto-detects and auto-configures the SIP servers that the provider configures (via TFTP or FTP of phone configuration files). All SIP traffic from the phones is directed at the servers operated by the provider, and not at the Aspen 365. Because the Aspen 365 also functions as a transparent layer 2 switch, it observes every SIP packet and decides whether to (a) forward or drop it (2) which ISP link is best suited for the specific phone s traffic (3) which SIP server is best suited for the specific phone, in the case redundant SIP servers are published, either via DNS SRV records or via SIP level re-directs. 2. VoIP Security Providers need to ensure (1) that their business customers IP phones are protected adequately from attack and fraud, and (2) that their own infrastructure is protected from attack and fraud if the customer premises PC network becomes compromised. The blends naturally into an environment where Data and traffic are placed on different s. Due to the transparent Layer 2 switch architecture of the Aspen 365, existing data firewalls are seamlessly accommodated while providing VoIP stateful firewall services. 2.1 Stateful Firewall for VoIP An optional VoIP firewall is included in the Aspen 365, to be used when a separate VoIP firewall does not exist.
6 In deploying new IP phones and a second private or link bundled with the VoIP service, the built in firewall in the Aspen 365 enables the following security benefits: 6 Automatic creation of a separate for, which ensures that any security breaches that might occur inside the Data are kept isolated from the A stateful firewall that enforces pin-holing i.e. tracks SIP exchanges and only opens up UDP/RTP ports based on a validated SIP state transition involving authorized SIP servers, and which closes these ports after the IP phone call has concluded. This stateful firewall ensures also that intrusion attempts and ports scans from outside are blocked and will fail. Security management with a built in DHCP server supporting multiple IP address pools, along with Network Address Translation (NAT) and Port Address Translation (PAT). Various flexible configurations are supported. In particular, existing best-of-breed Data Firewalls (which have all the state of the art application-layer protections for data) are supported. This is shown below in Fig 4. DSL Cable T1 DSL Cable T1 IP Phones use private IP space Switch Public IP address Best of breed data firewall Case 1: can provide VoIP firewall services to IP Phones while it co-exists with Customer s Existing Data Firewall. In this case, Aspen s convergence algorithms will ensure preferred treatment for streams IP Phones use private IP space Switch Public IP address Best of breed data firewall Case 2: Optionally, the customer s existing Data Firewall can bypass Aspen s convergence algorithms if desired, while ensuring multiple links for VoIP survivability Fig 4 Flexible VoIP Firewall Configurations 2.2 Network Admission Control for IP phones In conjunction with the Aspen DHCP function, network admission control can be enabled using either (1) the phone s MAC address, or (2) the phone s manufacturer ID (e.g. Polycom, Cisco).
7 7 Admission control is enforced by checking and verifying the MAC address of the device. By reserving a MAC address for a given IP address, no other devices are allowed the specific IP address. As an additional feature, it is possible to reserve IP addresses by specifying the 24 bit manufacturer id. This allows IP phones to have zero touch provisioning since each phone s MAC address does not have to be read and keyed in. 2.3 Preventing Theft of Service via SIP Auto Detect The Aspen 365 can be configured to auto-detect all SIP traffic, regardless of the SIP/UDP port being used. A SIP device can be classified as an intruder for such purposes, if it attempts to send out REGISTERs or INVITEs to the provider s server base. Various programmed alert messages can be sent to the provider s centralized monitoring system when such devices are detected. 2.4 Separate for IP Phones The Aspen VoIP firewall allows multiple DHCP pools. In order to create 2 separate s for voice and data, 2 pools with IP addresses from 2 different private IP subnets can be used. DSL Cable T1 IP Phones use private DHCP pool through Switch routing service plus NAT/PAT provided by Aspen 365. VoIP packets receive additional SIP ALG stateful firewall protection Desktop PCs use separate DHCP pool through Fig 5 Preventing Backdoor Attacks by Strict Separation First create a by using the phone manufacturer ID - and pick a pool such as through with router address , for example. This
8 8 feature is extremely valuable as it saves the hassle of knowing and/or configuring 48-bit Ethernet MAC addresses into the Aspen switch. For the PCs, the recommended practice is to assign them to a different subnet with or without DHCP. In the case of DHCP, as second DHCP pool can be used. If no separate data firewall exists, the Aspen 365 will still act as a NAT/PAT router for the PCs; this only provides minimal security for the PCs but does enforce a separation of traffic flows that is highly secure for the IP phones and the provider s VoIP infrastructure. If viruses or other malware compromises the PC network the IP phones remain secured. 2. WAN Optimization and Convergence The Aspen 365 is installed at the ideal point in the network to shape, control and optimize converged voice traffic on multiple links. Key Control Point: Aspen s SIP ALG can place voice traffic on different links on a phone-by-phone basis, based on real-time measured conditions. The traffic can be placed (1) on any link designated as enabled for voice and applies to BOTH incoming and outgoing phone calls, and (2) can be transparently re-directed at any SIP server on the basis of survivability considerations. The Aspen 365 monitors call quality via real-time measurements on voice RTP streams, then accumulates the statistics and feeds them back into the control loop. 3.1 Measurements and Statistics A few examples of statistics gathered by the are listed here: By processing SIP call signals, the Aspen 365 is able to gather information about specific IP phones (e.g. their call-ids, the codecs they use, and the bandwidth they require when active). By collecting statistics, important metrics such as the busy hour call volume and the duration of phone calls are tracked. The Aspen 365 is able to monitor the bandwidth demand from data applications such as web browsing, and video streaming. At the ISP link level, the Aspen 365 gathers information on available bandwidth and available bit rates, packet latency, packet loss as well as stream jitter. At the RTP stream level, the Aspen 365 is able to gather jitter, loss, latency and other statistics affecting call quality. 3.2 All IP VoIP and Cost Reduction The Aspen 365 provides tools and choices for the provider to design an All IP Phone network for each customer cost effectively. Correct design needs to take into account the size of the customer (how many IP phones are installed at each location), the bandwidth required by each active IP phone (depending on the voice codec, e.g. G.729 or G.711 etc), and the maximum number of concurrent users at any given time. Optimal design leads to lower costs, and naturally, higher operating margins.
9 9 At the lowest cost point, customers can provide their own broadband links both business cable and business DSL are at the point of user-level plug-and-play. The resulting simplicity in initial set up, and reductions in equipment and labor costs, all translate into a much lower cost of operations and ability to scale. These could be DSL links to two different ISPs or a mix of DSL, fixed wireless or Cable ISP for last mile path diversity. For small businesses with 10 to 25 users, in many cases this turns out to be an adequate and right-sized network design if analysis shows that the busy hour call volume never exceeds 8 or 9 concurrent phone calls. At the highest cost extreme, the provider can maintain a private T1 (or MPLS) network (and a private data center at the hub of this private network. However, operating such infrastructure is expensive for both the provider and the customer. An intermediate approach is to use standard T1 links to the, with dual T1s for fault tolerance; preferably each T1 is connected to a different ISP. A variation on this, which is also enabled, is 2 T1 links to the same ISP, but terminating in a different POP. All 3 cases are depicted with a cost graph in Fig 6 below. Cost 1 Bundled private lines High cost to operate infrastructure High cost to users Users Bundled T1 Lower cost to operate High cost for users Users Bring your own Broadband Lowest cost for operator and users Users Business Size Hosted IP PBX Provider Data Center T1 Private line or MPLS T1 Hosted IP PBX Provider Data Center T1 T1 Hosted IP PBX Provider Data Center DSL Cable Fig 6 Right-Sizing the Network In all cases, the Aspen 365 can load balance VoIP traffic when all links are UP, as well as fail over automatically when needed. In the case of heterogeneous links, such as DSL or cable the available bit rate on both uplink and downlink are taken into consideration in placing traffic. In examining the low end of the graph, where the largest market of small business users resides some key conclusions can be drawn. Since phone use is well known, most of these customers would not have a busy hour call volume that exceeds 8 or 9. Two business DSL links, or a mix of DSL and Cable, will cost only about one-fifth of the
10 10 monthly cost of dual T1 links. Typically that translates into a $600 per month cost saving for the customer! Further, the broadband routers are bundled in at no charge usually, and are also normally pre-set to plug-and-play enabling zero touch link provisioning. The flip side of this is the perceived risk of low-cost broadband. However, that is more than offset by the auto-correcting algorithms used in the Aspen 365 to compensate for instabilities, degrades and packet loss. 3.3 Convergence for and Data Optimization choices are flexible, and are largely based on what kinds of policies providers decide to implement or market to their business customers, and what kind of existing WAN links exist at customer premises, if at all, for data traffic. Various examples are discussed. The key to planning is to ensure sufficient bandwidth on the main and backup link for VoIP traffic, always. Case 1 Existing Cable Link for Data The provider can bring in a new DSL link for VoIP, assuming some sizing considerations are met. The existing cable ISP link can be used for voice backup if the DSL were to fail. In customer locations where the busy hour call volume never exceeds 8 calls, an ADSL link has sufficient capacity to handle VOIP traffic, see  for example. Since the G.729 codec can be used and only consumes about 33 Kbps per call, the total bandwidth needed for VoIP is about 270 kbps , well within the limited uplink capability of most ADSL links. Specific port-wiring and design considerations are depicted in Fig 7 below. DSL Cable Wire this to WAN port 2 for VoIP Wire this to WAN port 1 for Data Switch In the event the DSL link fails, VoIP would be recovered to the Cable ISP link, with a higher priority over the data traffic. The firewall must have a public IP address that is static on its untrusted port.
11 11 Fig 7 -Data Convergence Handling with Low Cost Broadband Asymmetric Fail Over is critical: If the data link fails (cable ISP link) the Aspen 365 will ensure that the data traffic does not fail over to the voice link (DSL). However, if the DSL link were to fail, the IP phone VoIP traffic will be recovered onto the cable ISP link. Note that in this case, the data firewall traffic will receive lower priority treatment using Aspen s convergence algorithms. Quality of Service and Traffic Shaping: During periods when the preferred voice link is down and the Aspen 365 has failed over voice to the DSL link, higher priority is given to voice traffic (RTP) over data traffic, ensuring that each stream receives adequate bandwidth. This will cause temporary degrade of performance for the data traffic. Variations on this solution can also be engineered to improve the situation for data traffic at times of VoIP link fail over. For example a total of 2 DSL links and 1 cable link (3 links in all) could be provisioned with data and voice balanced by the rich variety of VoIP/Data policy options offered in the Aspen 365. Case 2 Existing T1 for Data Certain types of fixed wireless, and high-speed cable are options for VoIP traffic when the customer/prospect has an existing T1 used for data traffic (se Fig 8 below). Another viable alternative is a second T1 for VoIP, but terminated at a different ISP. WISP 1/1 Mbps T1 Cable 6/1 Mbps T1 VoIP on Port 2 Data on Port 1 VoIP on Port 2 Data on Port 1 Switch Switch Case 1 Using a Fixed Wireless ISP (WISP) for VoIP 1 Mbps uplink, 1 Mbps Downlink Case 2 Using a premium Cable for Business ISP (WISP) for VoIP 1 Mbps uplink, 6 Mbps Downlink Fig 8 Right Sizing and Path Diversity Data uses existing T1
12 The Aspen 365 ensures Minimum disruption to data on the existing T1 during install and provisioning due to its optional wire mode and transparent bridge features. 2. Optimal converged voice management with the ability to support 50 to 100 IP phones and with busy hour call volumes in the range of 25 to Note that the solution also has robust last-mile diversity and survivability, something very essential in all-ip hosted IP PBX environments. 4. Serviceability and Provisioning The Aspen 365 has been engineered for ease of serviceability and IP phone provisioning by VoIP providers. The dual CPU architecture of the Aspen 365 includes a built in VoIP protocol analyzer tool, as well as real-time network measurement tools for jitter, packet loss and other metrics affecting call-quality. These can run concurrently without impacting the network performance of the Aspen 365 while under potentially heavy voice and data load. By decreasing mean time to repair (MTTR) and reducing instances of escalated intervention, providers can reduce overall costs while improving customer retention. For example: a problem as simple as a loose Ethernet cable attached to an IP phone can be discovered and rectified over the phone. 3.1 IP Phone Provisioning An optional feature for bulk provisioning is provided, whereby a reserved DHCP pool of IP addresses and an automatically created are made available to support first-time install procedures. An option to create an automatic based on the name of the phone manufacturer simplifies the process (since the need for the user or the provider s installer to type in the hardware MAC address is eliminated) 3.2 Secure Remote Management Secure remote access (via SSH, SCP) into the Aspen 365 management processor is provided; the packet forwarding processor is non-ip addressable and acts as a built-in firewall protecting the management processor from unauthorized access. Multiple levels of operations staff user can operate in different shells concurrently. An IP access control list can be used to limit access if desired to specific source computers or servers. Up to 8 different links can be used to access the appliance, either to troubleshoot an IP phone remotely or to push out configuration files in an automated manner. This ensures survivable remote access during various problem conditions that may afflict individual links. Simple problems such as an improperly connected or powered off device can be easily detected by first level technical support by remotely viewing the Aspen 365 management console.
13 Real-Time VoIP Protocol Tracing At times, a VoIP service request over the phone will require escalation to a protocolknowledgeable technician. The non-technical customer reporting the problem with phone service can communicate easily understandable information such as the phone (PSTN) number having the problem, or the SIP caller ID for the slightly more sophisticated. The Aspen 365 allows a technician to remotely start a real time trace of all IP packet activity on any such identified IP phone including protocol traces showing SIP and RTP state transitions. Various filters pertaining to SIP header fields are included, to ease the effort of isolating a specific phone. 3.4 Pro-Active Real Time Measurements The Aspen 365 packet forwarding processor is capable of continuously measuring the RTP payload in order to proactively measure: packet loss, round-trip time, payload receiver bit rate, jitter and other call-quality parameters. It does this as a background task on all VoIP calls in progress. This can be enabled as a proactive means of measuring call quality and detecting problems in advance of customer complaints. Alerts on various conditions can be generated to a centralized management console. 5. About the 365 Platform The ALG platform is built on dual processors for minimized voicejitter under high traffic conditions (since there may be a large number of PCs and phones in the network) SIP control packets and RTP voice streams are each handled on separate processors. Aspen s proprietary Aspen MOS is a fast-packet switching operating system optimized for deep-packet software inspection and handling of very intense packet loads. As is well known, VoIP streams use very small packets (typically about 100 bytes) and these result in relatively large, sustained packet-per-second rates in both directions. While the Forwarding processor timestamps, forwards and takes raw measurements on the voice stream, the Control processor processes SIP and SDP text based fields, performs statistical inferences on data and implements policy controls for traffic management in converged environments. References  Hosted VoIP: Steady Growth. But Will the Boom Come? In-Stat, David Lemelin, Analyst, December 2006  VoIP bandwidth -- Right size your WAN for Traffic, Robbie Harrell
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SSL-VPN Combined With Network Security Introducing A popular feature of the SonicWALL Aventail SSL VPN appliances is called End Point Control (EPC). This allows the administrator to define specific criteria
SIP Trunking Guide: Get More For Your Money 07/17/2014 WHITE PAPER Overview SIP trunking is the most affordable and flexible way to connect an IP PBX to the Public Switched Telephone Network (PSTN). SIP
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Beyond Quality of Service (QoS) Preparing Your Network for a Faster Voice over IP (VoIP)/ IP Telephony (IPT) Rollout with Lower Operating Costs Beyond Quality of Service (QoS) Cost Savings Unrealized THE
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Internet Security for Small to Medium Sized Businesses AN INTERNET SECURITY GUIDE FOR EVERY BUSINESS DOCUMENT VERSION 1.2 - OCTOBER 2002 COMPLIMENTS OF POWERWALLZ NETWORK SECURITY INC. 3 Introduction INTERNET
Aspen Solutions Branch Office IP Telephony: Improving the Reliability of Voice over Virtual Private Networks TECHNICAL NOTE Introduction IP PBX and IP phones are increasingly being deployed by small and
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In Brief Applications UC & VoIP Challenge Cut telecom cost by finding an alternative to costly MPLS Improve Internet connectivity speeds Fortify business continuity Solution Mushroom Networks Truffle Internet
Networking 4 Voice and Video over IP (VVoIP) Course Objectives This course will give delegates a good understanding of LANs, WANs and VVoIP (Voice and Video over IP). It is aimed at those who want to move
> White Paper Tough Questions, Honest Answers For many years, voice over IP (VoIP) has held the promise of enabling the next generation of voice communications within the enterprise. Unfortunately, its
A Light Reading Webinar Session Border Controllers in Enterprise Thursday, October 7, 2010 Hosted by Jim Hodges Senior Analyst Heavy Reading Sponsored by: Speakers Natasha Tamaskar VP Product Marketing
Note: This case study utilizes Packet Tracer. Please see the Chapter 5 Packet Tracer file located in Supplemental Materials. CHAPTER 5 OBJECTIVES Configure a router with an initial configuration. Use the
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Edgewater Routers User Guide For use with 8x8 Service Version 1.0, March 2011 Table of Contents EdgeMarc 200AE1-10 Router Overview...3 EdgeMarc 4550-15 Router Overview...4 Basic Setup of the 200AE1 and
ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. API: An application programming interface (API) is a source
Guide Cisco Virtual Office Unified Contact Center Architecture Contents Scope of Document... 1 Introduction... 1 Platforms and Images... 2 Deployment Options for Cisco Unified Contact Center with Cisco
Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents First Steps... 3 Identifying your MAC Address... 3 Identifying your Dynamic IP Address...
Remote Monitoring Software for Managed Services Providers PacketTrap RMM provides a cost-effective way for you to offer enterprise-class server, application, and network management to your customers. It
UIP1868P User Interface Guide (Firmware version 0.13.4 and later) V1.1 Monday, July 8, 2005 Table of Contents Opening the UIP1868P's Configuration Utility... 3 Connecting to Your Broadband Modem... 4 Setting
Achieving Service Quality and Availability Using Cisco Unified Communications Management Suite EXECUTIVE SUMMARY CISCOLIVE Europe 2010 Annual Cisco IT and communications conference Event held at Barcelona,
An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this
Virtual Leased Line (VLL) for Enterprise to Branch Office Communications Reliable high throughput data connections with low-cost & diverse transport technologies Executive Summary: The Truffle Broadband
Whitepaper WAN Traffic Management with PowerLink Pro100 Overview In today s Internet marketplace, optimizing online presence is crucial for business success. Wan/ISP link failover and traffic management
Enterprise SIP Designed For Market Requirements Enterprises can combine XO Enterprise SIP with ANY MPLS IP-VPN or Data Network (even from another carrier) for an all-in-one, multi-site IP communications
SIP Security Controllers Product Overview Document Version: V1.1 Date: October 2008 1. Introduction UM Labs have developed a range of perimeter security gateways for VoIP and other applications running
Break Internet Bandwidth Limits Higher Speed. Extreme Reliability. Reduced Cost. Peplink. All Rights Reserved. Unauthorized Reproduction Prohibited Presentation Agenda Peplink Balance Pepwave MAX Features