1. 1 VoIP Fundamentals Tech 160

2. 2 Agenda Tech 160  Voice Communication  Voice over PSTN  Voice over IP  Quality of Service  VoIP Security  References

3. 3 Voice Communication The Traditional Way – Voice over PSTN Migration from the entirely analogue PSTN to the digital technology starting in 1975 yielded many advantages such as:  Greater bandwidth  Reduced error rates  Enhanced management and control For converting the analogue signal into a digital format and reconvert it back into an analogue one, a matching pair of codecs (coder/decoder) is used (PCM = Pulse Code Modulation, standardized by the ITU-T as G.711).

4. 4 Voice over PSTN Required Bandwidth for Standard Packets: Required Bandwidth for Encrypted Packets (in the case that a VPN is used):

5. 5 Voice over PSTN PSTN Advantages  Reliable  Good voice quality and minimal delays  Worldwide phone coverage PSTN Drawbacks  Inefficient use of the available bandwidth  Costly long distance service Worldwide phone coverage  Complex provisioning of the available services

6. 6 VoIP Definition: Voice over IP (VoIP), also called Internet telephony or IP telephony, is The transmission of voice telephony services over IP, the Internet Protocol. It is a general term that refers to any means of converting voice calls into voice data packets that are transmitted over an IP network, either public or private.

7. 7 VoIP Timing Because IP does not, by default, provide any QoS mechanism, parameters or limits had to be placed on latency, jitter and loss in order to achieve acceptable voice quality. Compression VoIP often uses DSP that compress and process the signal, thereby requiring less than the 64 Kbps required by PCM. The result is encapsulated inside an IP packet, along with a UDP header for purposes of multiplexing, header error control, and identification of the application by port number. RTP is run for end-to-end delivery services such as payload type identification, packet sequence numbering, time stamping, and delivery monitoring. On the receiving end, these Internet Protocol processes are then reversed and the voice is extracted and played to the listener.

8. 8 VoIP Packet Loss, Latency and Jitter A major complication of VoIP is that these packets are not delivered at the same pace they entered the network. Additionally, some packets may be lost in transit, due to the IP network being a highly shared packet network characterized by unpredictable levels of congestion. There is variability in the latency, jitter and loss of the packets. The use of intelligent algorithms is applied to resolve these packet issues to minimize their impact on the quality of voice. These algorithms are designed to fill in the voids by stretching the voice frames received earlier and blending them with those received later. Echo Cancellation VoIP also makes use of various techniques for echo cancellation, as echo becomes perceptible when delays exceed 15ms-20ms.

9. 9 VoIP VoIP Advantages: Use of the same network for data and voice resulting in:  Reduced cabling and infrastructure costs  Simplified network administration and control  Voice and data staff merge  Productivity and service delivery improvements Highly cost-effective and possibly free long distance telephony compared to PSTN (depending on whether a public or a private PSTN/VoIP gateway is used) Means to consolidate and unify phone services and features from different locations Future feature enhancement possibilities

10. 10 VoIP VoIP Drawbacks:  Some loss of voice data under heavy data traffic due to traffic volume unpredictability over the Internet, resulting in packet loss, delays or jitter. For the majority of the connections this will not be experienced.  Some countries may regulate the use of VoIP to protect local PSTN monopolies, which could lead to the introduction of VoIP call fees over the public Internet. Currently there are no specific rates established by any country.  Limited gateway coverage to bridge the PSTN with the data network in order to reach any telephone around the world. This limitation is non-existent if the VoIP device has PSTN connectivity in addition to VoIP capability.

11. 11 Quality of Service Quality of Service (QoS) In packet-switched networks, quality of service is a mechanism used to prioritize network traffic to ensure that network devices handle high-priority traffic first. In VoIP, QoS mechanisms guarantee that the values for packet latency, jitter, loss of packets and other parameters affecting the voice quality are within an acceptable range. Service Level Agreement (SLA) Service-level agreement is an agreement between a customer and a service provider that guarantees basic performance benchmarks, usually in exchange for a commitment to spend a fixed amount with the provider over a time period. Some metrics that SLAs may specify include: the percentage of the time services will be available, the number of users that can be served simultaneously, etc.

12. 12 Quality of Service MoS (Mean Opinion Score) VoIP service SLAs tend to be judged on their Mean Opinion Scores (MOS). They can include such factors as call completion rates and the length of time required for a user to hear a dial tone or to connect to the dialed party. Various measurement techniques are used in association with SLAs, including active network tests made at regular time intervals as well as passive measurements that are based on actual calls placed across network.

13. 13 VoIP Signaling Protocols As packet telephony networks grew in the late 90s and interconnection dependencies emerged, it became clear that the industry needed standard VoIP protocols. Several groups developed independent standards, each with its own unique characteristics. There are four main VoIP call-control protocols (also known as signaling protocols):  Session Initiation Protocol (SIP) is an RFC standard (RFC 3261) from the Internet Engineering Task Force(IETF)  H.323 (an ITU recommendation defining "packet-based multimedia communications systems")  Media Gateway Control Protocol (MGCP)  H.248/Megaco (an ITU recommendation defining a "Gateway Control Protocol")

14. 14 SIP The Session Initiation Protocol (SIP)  Is a signalling protocol used for establishing sessions in an IP network. A simple session could be a two-way telephone call or it could be a video and messaging communication between two devices.  Is a request-response protocol that closely resembles two other Internet protocols, HTTP and SMTP (the protocols that power the world wide web and email); consequently, SIP sits comfortably alongside Internet applications. There are two basic components within SIP:  The SIP user agent = the end system component for the call  The SIP network server = the network device that handles the signaling associated with multiple calls. SIP is implemented by installing servers on the data network, either private or on the public internet, that run SIP software that allows users to register the location (in the form of an IP address) of their phone or device. The SIP address for a device is similar to an email address and takes the form user@location.com (i.e. 30112345@sip.epygi.com). Once registered on the SIP server users can call other registered users.

15. 15 VoIP Security VoIP security is a major issue in VoIP communication and should never be underestimated. Fraudulent users can easily take advantage of an insecure VoIP system and place calls, causing extra traffic and toll charges. Eavesdropping is another threat to the security of VoIP communication. VoIP Security Tips  VPN: Run VoIP traffic over a VPN.  Secure RTP (SRTP): Use SRTP to encrypt RTP payload.  Secure SIP (SSIP): Use SSIP to encrypt the SIP messaging port.  Firewall: Allow traffic from trusted network addresses only.  Call Filtering: Allow calls from trusted numbers only.  Authentication: Authenticate users prior to providing VoIP services.  Quadro System Security Management: Quadro SIP IDS and Security Audit

16. 16 SIP Resources on the Internet: General Information Sites:  http://www.voip-info.org/tiki-index.php VoIP Service Provider List:  http://www.voipproviderslist.com/ SIP Training and Certifications:  http://www.thesipschool.com/ Free news and magazines:  http://www.tmcnet.com/ Internet Engineering Task Force:  http://www.ietf.org/  http://www.ietf.org/rfc/rfc3550.txt (RTP- Real Time Protocol)  http://www.ietf.org/html.charters/sip -charter.html (SIP) International Telecommunication Union (ITU):  http://www.itu.int/home/index.html

17. 17 The end