Vo ip overview
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This document provides an overview of Voice over Internet Protocol (VoIP) technology. It discusses key components of VoIP including signaling protocols, audio codecs, echo elimination, advantages, applications, and considerations for deployment. The document contains slides that cover topics such as the public switched telephone network versus VoIP networks, signaling systems, audio codecs, quality of service factors, security threats, integrating VoIP with existing phone systems, and virtualized PBX solutions.
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Vo ip overview
- 1. VoIP Overview Viroj Tanggasemsun CCOP Co., Ltd.
- 2. 07/21/15 slide 2 Overview • Technology Introduction • Signaling System • Audio Codec • Echo • Gaining from VoIP • Advantages • Killer Applications • Getting started with VoIP • Key components of the system • Concerning aspects when deploying VoIP • Case Studies
- 3. 07/21/15 slide 3 Voice Network • Public Switched Telephone Network (PSTN) • A network for voice communication bases on circuit-switched connection. • PSTN originally uses for analog telephone system, but now also uses for digital telephone system. • Voice over Internet Protocol (VoIP) • Technology for delivery of voice over IP network which provides multiple paths from source to destination.
- 4. 07/21/15 slide 4 Signaling System 7 (SS7) • Local-Loop signaling • Supervisory signaling • Address signaling • Information signaling • Trunk signaling provides information for setting up and teardown a voice channel between a pair of telephone exchange.
- 5. 07/21/15 slide 5 Signaling Protocols for VoIP • Many applications of the Internet require the creation and management of a session. • The signaling protocols were developed to control the access methods and sessions between two or more end-points. • VoIP signaling protocols • Media Gateway Control Protocol (MGCP & MEGACO) • H.323 • Session Initiation Protocol (SIP)
- 6. 07/21/15 slide 6 Audio Codec • Non-compression: WAV • Lossless data compression: ALAC, Blu-ray • Lossy data compression: Dolby Digital (AC3), MPEG, ITU standards (G.xxx) Codec Bit Rate (Kbps) G.711 64 G.722 48, 56, 64 G.726 (G.721) 16, 24, 32 G.728 16 G.729 8
- 7. 07/21/15 slide 7 Eliminating an echo • Echo is a sound of speaker voice being played back to the speaker after a delay. • Hybrid Echo • Acoustic Echo • Eliminating an echo • Acoustic echo control is based on non-linear filtering. The related standard is G.160. • Echo suppression is work by turning off the receive side when speaker is transmitting. • Echo cancelation is normally implemented using digital signal processing technique.
- 8. 07/21/15 slide 8 Advantages of using VoIP • Network Infrastructure • only an existing data network is needed to serve both voice and data traffic • integration with other networks using gateway • Services • user-location independence • integration with other services available over the Internet such as e-mail, instance messaging • many novel services can be provided. • Cost • no additional cost is provided for separate voice and data networks. • there is only cost of broadband connection without billing per-call or per-minute.
- 9. 07/21/15 slide 9 Disadvantages of using VoIP • Complicated network architecture and interoperability between different protocols. • Ensuring quality of service is difficult. • Security issues of voice over IP are much more than line intercepting in legacy system. • Service not available during a power outage. • Tracking user location is not available for emergency call.
- 10. 07/21/15 slide 10 Killer Application • PC-to-Phone allows making a call from your computer to traditional telephone. • Phone-to-PC lets your computer has its own number for calling from traditional telephone. • Web Call or Click-to-Dial controls your call via a web. • Web Conference let you simply attend a conference via a web.
- 11. 07/21/15 slide 11 Killer Application II • Push-to-Talk turns a mobile phone into a Walkie-Talkie. • Unified Communications and unified messaging combine any media services into the same session or same box. • VoIP over WLAN/Wi-Fi/3G/WiMAX together with the device that can automatically select a suitable network make cost effective for your communications.
- 12. 07/21/15 slide 12 Unified Communication
- 13. 07/21/15 slide 13 Technical Aspects of Deploying VoIP • Overall performance of the network must be evaluated to make sure the new VoIP application can run smoothly. • The parameters such as loss, delay, and jitter must be concerned for the quality of the VoIP. • VoIP Security is an important issue to rise the confidence of the users.
- 14. 07/21/15 slide 14 Overall Performance • Bandwidth consumption mainly depends on the CODEC used. CODEC Bit Rate (Kbps) Bandwidth on Frame Relay (Kbps) Bandwidth of Ethernet (Kbps) G.711 G.728 G.729 64 16 8 67.6 18.4 11.6 87.2 31.5 31.2 IP UDP RTP Payload Header 40 bytes Ethernet frame of voice packet
- 15. 07/21/15 slide 15 Overall Performance II • The system must be designed for a high availability and reliability. • Redundant server must be implemented (Resiliency). • Prepare a solution for power outage problem.
- 16. 07/21/15 slide 16 Voice Quality • There are three indicators that can measure quality of voice communications over IP network. • Packet loss • End-to-end delay • Jitter • The quality of equipments such as the server and the IP Phone.
- 17. 07/21/15 slide 17 Assessing Quality of VoIP • Mean Opinion Score (MOS) is measured from an opinion of a large number of people who listen to the voice. • R-factor is derived from loss, delay, jitter and equipment impairment factors. Very Satisfied Satisfied Some Users Dissatisfied Many Users Dissatisfied Nearly All Users Dissatisfied Not Recommended 100 94 90 80 70 60 50 0 R 4.4 4.3 4.0 3.6 3.1 2.6 1.0 MOS G.107 Default Value EstimatedMOS One Way Delay (ms) 0 100 200 300 400 500 2.5 3 3.5 4 4.5 5 G.711 G.729 G.723-MPMLQ G.723-ACELP G.726
- 18. 07/21/15 slide 18 Voice over IP Security Threats • DoS attacks and registration flooding are threat that can stop the service of the system. • Sniffing a voice signaling and media is a threat against user confidentiality. • Call spam or SPIT is a threat like spam-mail which is against social context. Solution Session Border Control (SBC) AES encryption Media / Voice PSTN Call Control TCP/IP Network Manage ment Policy
- 19. 07/21/15 slide 19 Integrating with existing PBX IP Network PSTN SIP trunk SIP trunk Existing PBX Existing LAN switch New LAN switch
- 20. 07/21/15 slide 20 Connecting between Multi sites 3300 ICP (Integrated) PSTN Headquarters 3300 ICP (Media Gateways) PSTN Hosted Apps Centralized Management 3300 ICP (Call Controller) Branch Office 3300 ICP (Media Gateway)PSTN Small Office Small Branch or Home Office Line Interface Module DSL / Cable Modem IP SetLAN PSTN WAN / Internet
- 21. 07/21/15 slide 21 Security Implementation
- 23. 07/21/15 slide 23 New Trend of Virtualized PBX
- 24. 07/21/15 slide 24 Virtual Application UCaaS
- 25. 07/21/15 slide 25 The information conveyed in this presentation, including oral comments and written materials, is confidential and proprietary to Mitel and is intended solely for Mitel® employees and members of Mitel’s reseller channel. If you are not a Mitel employee or a Mitel reseller, you are not the intended recipient of this information and are not invited to the conference, and cannot participate in or listen to and/or view the presentation. Please delete or return any related material. Mitel will enforce its rights to protect its confidential and proprietary information, and failure to comply with the foregoing may result in legal action against you or your company.
Editor's Notes
- Signaling for Analog Telephone Networks: In a switched telephone network, signaling conveys the intelligence needed for one subscriber to interconnect with any other in that network. Signaling tells the switch that a subscriber desires service and then gives the local switch the data necessary to identify the required distant subscriber and hence to route the call properly. It also provides supervision of the call along its path. Signaling also gives the subscriber certain status information, such as dial tone, busy tone(busy back), and ringing. Metering pulses for call charging may also be considered a form of signaling. There are several classifications of signaling: General Subscriber signaling (Local-loop signaling). Interswitch signaling (Trunk signaling). Functional Supervisory. Address signaling. Audible-visual (call progress and altering). Signaling functions _____________________________________________|__________________________________________ | | | Supervisory Address Audible-visual _______|________ ______|_______ _______|________ | | | | | | Control(forward) Status(backward) Station Routing Altering Progress - Seize - idle - rotary dial - channel - ringing - dial tone - Hold - busy - push button - trunk - paging - busy tone - Release - disconnect - digital - off-hook warning - ring back It should be appreciated that on many telephone calls, more than one switch is involved in call routing. Therefore switches must interchange information among switches in fully automatic service. Address information is provided between modern switching machines by inter-register signaling, and the supervisory function is provided by line signaling. The audible-visual category of signaling functions inform the calling subscriber regarding call progress. The altering function informs the called subscriber of a call waiting or an extended “off-hook” condition of his or her handset. Signaling information can be conveyed by a number of means from subscriber to switch or between (and among) switches. Signaling information can be transmitted by means such as: Duration of pulses (pulse duration bears a specific meaning), Combination of pulses, Frequency of signal, Combination of frequencies, Presence or absence of a signal, binary code. Signaling System 7- On the public switched telephone network (PSTN), Signaling System 7 (SS7) is a system that puts the information required to set up and manage telephone calls in a separate network rather than within the same network that the telephone call is made on. Signaling information is in the form of digital packets. SS7 uses what is called out-of-band signaling, meaning that signaling (control) information travels on a separate, dedicated 56 or 64 Kbps channel rather than within the same channel as the telephone call. Historically, the signaling for a telephone call has used the same voice circuit that the telephone call traveled on (this is known as in-band signaling). Using SS7, telephone calls can be set up more efficiently and with greater security. Special services such as call forwarding and wireless roaming service are easier to add and manage. SS7 is now an international telecommunications standard. SS7 is used for these and other services: Setting up and managing the connection for a call Tearing down the connection when the call is complete Billing Managing call forwarding, calling party name and number display, three-way calling, and other Intelligent Network (IN) services Toll-free (800 and 888) and toll (900) calls Wireless as well as wireline call service including mobile telephone subscriber authentication, personal communication service (PCS), and roaming SS7 messages contain such information as: How should I route a call to 914 331-4985? The route to network point 587 is crowded. Use this route only for calls of priority 2 or higher. Subscriber so-and-so is a valid wireless subscriber. Continue with setting up the call. Because control signals travel in a separate network from the call itself, it is more difficult for anyone to violate the security of the system. (See 2600 and phreak for cracking techniques that are defeated by SS7.) The Integrated Services Digital Network (ISDN) also uses out-of-band signaling, extending it all the way to the end user on the ISDN D-channel while voice and data flow on B channels. Briefly How It Works SS7 consists of a set of reserved or dedicated channel known as signaling links and the network points that they interconnect. There are three kinds of network points (which are called signaling points): Service Switching Points (SSPs), Signal Transfer Points (STPs), and Service Control Points (SCPs). SSPs originate or terminate a call and communicate on the SS7 network with SCPs to determine how to route a call or set up and manage some special feature. Traffic on the SS7 network is routed by packet switches called STPs. SCPs and STPs are usually mated so that service can continue if one network point fails. SIGTRAN- SIGTRAN (for Signaling Transport) is the standard Telephony protocol used to transport Signaling System 7 (SS7) signals over the Internet. SS7 signals consist of special commands for handling a telephone call. Internet telephony uses the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Calls transmitted over the Internet travel as packets of data on shared lines, avoiding the tolls of PSTN. A telephone company switch transmits SS7 signals to a signaling gateway. The gateway, in turn, converts the signals into SIGTRAN packets for transmission over IP to either the next signaling gateway or, if the packet destination is not another PSTN, to a softswitch. The SIGTRAN protocol is actually made up of several components (this is what is sometimes referred to as a protocol stack): standard IP; a common signaling transport protocol (used to ensure that the data required for signaling is delivered properly), such as the Stream Control Transport Protocol (SCTP); and an adaptation protocol that supports "primitives" (a basic interface or segment of code that can be used to build more sophisticated program elements or interfaces) that are required by another protocol.
- SIP System to PSTN Interconnection This figure shows the SS7 that is used to initiate a call into a SIP system. This diagram shows that a caller that is connected to a SIP network (SIP client) initiates a call using an invite command that contains a destination SIP Uniform Resource Locator (URL). This identifier is sent to the proxy server that determines and maps the URL to the actual destination number that will be sent in the initial address message (IAM). The proxy server informs the SIP client that the call routing is in progress (it is trying to connect). The Invite command is then forwarded to the network gateway (NGW). The NGW creates an IAM that contains the destination phone number. The PSTN switch sends back an ACM to the NGW. The NGW informs the proxy server that the call is progressing and the proxy server forwards this session progress message to the SIP client. This allows an audio path to be connected between the PSTN and the SIP client. When the destination telephone user answers, the PSTN sends and answer message to the NGW. The NGW translates this command and sends a message updating the session to indicate the call has been answered. This is forwarded to the SIP client. When the SIP client acknowledges the message, the NGW can connect a second media path from the SIP client to the PSTN switch.
- WAV From Wikipedia, the free encyclopedia: Waveform Audio File Format (WAVE/WAV)Filename extension.wav .waveInternet media typeaudio/vnd.wave,[1] audio/wav, audio/wave, audio/x-wav[2]Type codeWAVEUniform Type Identifiercom.microsoft.waveform-audioDeveloped byMicrosoft & IBMInitial release1991 (1991)[3]Latest releaseMultiple Channel Audio Data and WAVE Files / 7 March 2007; 7 years ago (2007-03-07) (update)[4][5]Type of formataudio file format, container formatExtended fromRIFFExtended toBWF, RF64Waveform Audio File Format (WAVE, or more commonly known as WAV due to its filename extension)[3][6][7][8] (rarely, Audio for Windows[9]) is a Microsoft and IBM audio file format standard for storing an audio bitstream on PCs. It is an application of the Resource Interchange File Format (RIFF) bitstream format method for storing data in "chunks", and thus is also close to the 8SVX and the AIFF format used on Amiga and Macintosh computers, respectively. It is the main format used on Windows systems for raw and typically uncompressed audio. The usual bitstream encoding is the linear pulse-code modulation (LPCM) format. Apple Lossless From Wikipedia, the free encyclopedia: Apple LosslessDeveloper(s)Apple Inc.Initial releaseApril 28, 2004; 10 years ago (2004-04-28)Stable releaseOctober 28, 2011; 2 years ago (2011-10-28)TypeAudio codecLicenseApache License 2.0Websitealac.macosforge.orgFilename extension.m4aDeveloped byApple Inc.Type of formatLossless data compression, audio file formatContained byMPEG-4 Part 14Apple Lossless, also known as Apple Lossless Audio Codec (ALAC), or Apple Lossless Encoder (ALE), is an audio codec developed by Apple Inc. for lossless data compression of digital music. After initially keeping it proprietary from its inception in 2004, in late 2011 Apple made the codec available open source and royalty-free. Traditionally, Apple has referred to the codec as Apple Lossless, though more recently they have begun to use the abbreviated term ALAC when referring to the codec.[1]
- Acoustic echo arises when sound from a loudspeaker—for example, the earpiece of a telephone handset—is picked up by the microphone in the same room—for example, the mic in the very same handset. The problem exists in any communications scenario where there is a speaker and a microphone. Hybrid echo is generated by the public switched telephone network (PSTN) through the reflection of electrical energy by a device called a hybrid (hence the term hybrid echo). Most telephone local loops are two-wire circuits while transmission facilities are four-wire circuits. Each hybrid produces echoes in both directions, though the far end echo is usually a greater problem for voiceband.
- The amount of bandwidth required to carry voice over an IP network is dependent upon a number of factors. Among the most important are: ¯ Codec (coder/decoder) and sample period ¯ IP header ¯ Transmission medium ¯ Silence suppression The Codec: The codec determines the actual amount of bandwidth that the voice data will occupy. It also determines the rate at which the voice is sampled. The conversion of the analogue waveform to a digital form is carried out by a codec. The codec samples the waveform at regular intervals and generates a value for each sample. A sample period of 20 ms is common. a G.711 codec sampling at 20 ms. This generates 50 frames of data per second. G.711 transmits 64,000 bits per second so each frame will contain 64,000/50 = 1,280 bits or 160 octets. Some codecs use longer sample periods, such as 30 ms employed by G.723.1. Others use shorter periods, such as 10 ms employed by G.729a Frames and Packets: Many IP phones simply place one frame of data in each packet. However, some place more than one frame in each packet. For example, the G.729a codec works with a 10 ms sample period and produces a very small frame (10 bytes). It is more efficient to place two frames in each packet. This decreases the packet transmission overhead without increasing the latency excessively. Latency and Packet Overhead: Long sample periods produce high latency, which can affect the perceived quality of the call. Long delays make interactive conversations awkward, with the two parties often talking over each other. Based on this fact alone, the shorter the sample period, the better the perceived quality of the call. However, there is a price to pay. The shorter the sample period, the smaller the frames and the more significant the packet headers become. For the smallest packets, well over half of the bandwidth used is taken up by the packet headers The term ‘IP header’ is used to refer to the combined IP, UDP and RTP information placed in the packet. The payload generated by the codec is wrapped in successive layers of information in order to deliver it to its destination. These layers are: IP – Internet Protocol UDP – User Datagram Protocol RTP – Real-time Transport Protocol RTP is the first, or innermost, layer added. This is 12 octets. RTP allows the samples to be reconstructed in the correct order and provides a mechanism for measuring delay and jitter. UDP adds 8 octets, and routes the data to the correct destination port. It is a connectionless protocol and does not provide any sequence information or guarantee of delivery. IP adds 20 octets, and is responsible for delivering the data to the destination host. It is connectionless and does not guarantee delivery or that packets will arrive in the same order they were sent. In total, the IP/UDP/RTP headers add a fixed 40 octets to the payload. With a sample period of 20 ms, the IP headers will generate an additional fixed 16 kbps to whatever codec is being used. The payload for the G.711 codec and 20 ms sample period calculated above is 160 octets, the IP header adds 40 octets. This means 200 octets, or 1,600 bits sent 50 times a second– result 80,000 bits per second. This is the bandwidth needed to transport the Voice over IP only, it does not take into account the physical transmission medium The Transmission Medium: In order to travel through the IP network, the IP packet is wrapped in another layer by the physical transmission medium. Most Voice over IP transmissions will probably start their journey over Ethernet, and parts of the core transmission network are also likely to be Ethernet. Ethernet has a minimum payload size of 46 octets. Carrying IP packets with a fixed IP header of 40 means that the codec data must be at least 6 octets – typically not a problem. The Ethernet packet starts with an 8 octet preamble followed by a header made up of 14 octets defining the source and destination MAC addresses and the length. The payload is followed by a 4 octet CRC. Finally, the packets must be separated by a minimum 12 octet gap. The result is an additional Ethernet overhead of 38 octets. Ethernet adds a further 38 octets to our 200 octets of G.711 codec frame and IP header. Sent 50 times a second – result 95,200 bits per second, see example 1 below. This is the bandwidth needed to transmit Voice over IP over Ethernet. Transmission of IP over other mediums will result in different overhead calculations. Voice over IP over Ethernet, Example 1: G.711 ¯ Codec G.711 – 64 kbps, 20 ms sample period ¯ 1 frames per packet (20 ms) ¯ Standard IP headers ¯ Ethernet transmission medium One packet is sent every 20 ms, 50 packets per second. Payload is 64,000 ÷ 50 = 1,280 bits (160 octets). Fixed IP overhead 40 octets, fixed Ethernet overhead 38 octets. Total size 238 octets. Bandwidth required is (160 + 40 + 38) x 50 x 8 = 95,200 kbps. Voice over IP over Ethernet, Example 2: G729a ¯ Codec G.729a – 8 kbps, 10 ms sample period ¯ 2 frames per packet (20 ms) ¯ Standard IP headers ¯ Ethernet transmission medium One packet is sent every 20 ms, 50 packets per second. Payload is 8,000 ÷ 50 = 160 bits (20 octets). Fixed IP overhead 40 octets, fixed Ethernet overhead 38 octets. Total size 98 octets. Bandwidth required is (20 + 40 + 38) x 50 x 8 = 39,200 kbps. Silence Suppression: Certain codecs support silence suppression. Voice Activity Detection (VAD) suppresses the transmission of data during silence periods. As only one person normally speaks at a time, this can reduce the demand for bandwidth by as much as 50 percent. The receiving codec will normally generate comfort noise during the silence periods.
- Resiliency on the 3300 ICP increases communications reliability by maintaining calls in progress, handling new incoming and outgoing calls, and continuing to provide voice mail services in the event of 3300 ICP or network failure. Advantages Over Redundancy Resiliency is less costly and more flexible than a redundant solution, because it uses self-correction techniques that take advantage of the IP network characteristics of location independence and network element distribution. While the redundancy model is highly effective and reliable, it is unnecessarily costly for some customers.
- คุณภาพของเครือข่ายและคุณภาพเสียงสำหรับ VoIP สิ่งหนึ่งที่จำเป็นต้องพิจารณาสำหรับการเปลี่ยนมาใช้ VoIP คือ ความสามารถของระบบเครือข่ายในการส่งแพ็กเก็ตข้อมูลได้อย่างทันเวลา ไม่สะดุด ซึ่ง TCP/IP ไม่ได้รับประกันว่าทุกแพ็กเก็ตข้อมูลที่ส่งออกไปจะไปถึงจุดหมายปลายทางเสมอ แม้แต่เราเตอร์เอง ก็สามารถปรับตั้งค่าให้ละทิ้งแพ็กเก็ตส่วนเกินเพื่อบรรเทาปัญหาการคับคั่งของแพ็กเก็ต (packet congestion) หรืออุปสรรคอื่น ๆ อีก ซึ่งพอสรุปได้ดังนี้ • Packet Lost โดยการวัดจากเปอร์เซ็นต์ของแพ็กเก็ตที่ไม่ถึงจุดหมายปลายทาง หากสูงกว่า 3% ถือว่าเครือข่ายนั้นไม่เหมาะสมกับการใช้งาน VoIP เนื่องจากอาจจะมีปัญหาสัญญาณเสียงขาดตอน ซึ่งปัญหา packet loss นี้ จะเพิ่มมากขึ้นตามลักษณะการใช้งานของระบบเครือข่ายที่เพิ่มสูงขึ้น โดยเฉพาะจุดที่มีการ overload ของสัญญาณ • Jitter คือประเด็นด้านคุณภาพของระบบเครือข่ายที่วัดจากความแปรปรวนของเวลาที่ใช้ในการเดินทางของแพ็กเก็ตที่เป็นสัญญาณเสียง (Voice Information Packet) ซึ่งอุปกรณ์ VoIP ที่ฝั่งผู้รับสามารถบรรเทาปัญหานี้ ได้ด้วยการจัดให้แพ็กเก็ตที่ได้รับมารวมตัวกันอยู่ใน jitter buffer ก่อนแปลงเป็นสัญญาณเสียงที่ไม่มีการสะดุดหรือขาดตอน Jitter Buffer มีค่าความยาวเป็นมิลลิวินาที (Millisecond) หรือที่เรียกว่า Jitter Buffer Depth ซึ่งควรมีค่าประมาณ 2 เท่าของขนาดของความแปรปรวนของระยะเวลาในการเดินทางของแพ็กเก็ตหรือ jitter ที่เกิดขึ้นจริงในระบบเครือข่าย หากค่า jitter มากกว่า 50 มิลลิวินาทีแล้ว จะเป็นการยากที่จะได้สัญญาณเสียงที่ราบเรียบ อีกทั้งการใช้ jitter buffer บ่อยครั้ง ก็จะยังผลให้เกิดอาการสัญญาณเสียงขาดหายเป็นช่วง ๆ ดังนั้นการปรับตั้งค่า Jitter Buffer Depth จึงต้องมีความเหมาะสมกับลักษณะของระบบเครือข่ายด้วย • Latency คือระยะเวลาที่ใช้ในการเดินทางของแพ็กเก็ตจากต้นทางไปยังปลายทาง หากใช้เวลามากกว่า 150-200 มิลลิวินาที อาจเป็นปัญหาสำหรับอุปกรณ์ VoIP ได้ในรูปของเสียงสะท้อนหรือเสียงก้อง และหากมีค่ามากกว่า 400 มิลลิวินาทีอาจมีผลกระทบต่อความชัดเจนของเสียงสนทนา แต่ทั้งนี้หากการสื่อสารของทั้งระบบเป็นอุปกรณ์แบบดิจิตอลทั้งหมด ปัญหาเรื่องเสียงก้องหรือเสียงสะท้อนก็จะน้อยมาก
- Media Path (or Voice Stream) The audio and video streams between communicating endpoints Signalling Path (or Call Control) Signalling between Call Controllers and phones as well as applications Management Path Access to system data for the purpose of configuration, provisioning, retrieval of performance data and other system logging information Both local access and remote access. Both Human access (UIs) and Program access (APIs) PSTN and Legacy Devices Connections to external TDM communications systems TCP/IP Network The core network must be secure for basic TCP/IP traffic Policy