SIP Trunking and Voice over IP

Transcription

1 SIP Trunking and Voice over IP

2 Agenda What is SIP Trunking? SIP Signaling How is Voice encoded and transported? What are the Voice over IP Impairments? How is Voice Quality measured? VoIP Technology Confidential & Proprietary Information of VeEX Inc. 2

3 Speech Encoding Speech encoding is used to compress and encode human speech before it is transmitted through the network. Voice Sampling 1 0 Analog Voice Signal Digitized Voice Signal Digital Signal transmission For the G.711 (PCM) codec, the voice is sampled at a rate of 8 khz (8,000 samples per second) and digitized using 8-bit samples before being transmitted VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 3

4 Useful Definitions TRUNK: A trunk is a circuit that connects telephone switches. Private Branch Exchange (PBX): PBXs make connections among the internal telephones of a private organization and also connect them to the via trunk lines Trunking saves cost, because there are usually fewer trunk lines than extension lines, since it is unusual in most offices to have all extension lines in use for external calls at once PSTN Trunk Line PBX Extensions

5 Time Division Multiplexing (TDM) T1 The T1 signal takes 24 DS0 channels and multiplexes them into a single bit stream using Time Division Multiplexing The multiplexing is octet (8 bits) oriented. 8 bits for each channel are contiguous. Each channel uses 64kbps, the bandwidth is reserved for each channel whether there is an active call or not. : : : :8bits : 8bits 8bits8bits : Timeslot 0 Timeslot 1 Timeslot 23 Timeslot 0 T1 and ISDN Training Confidential & Proprietary Information of VeEX Inc. 5

6 ISDN PRI ISDN = Integrated Services Digital Network With ISDN voice and data can be carried simultaneously B Channel (Bearer) = 64 kbps for voice D Channel (Data) = 16 or 64 kbps for signaling or data ISDN Services: ISDN PRI: Primary Rate Interface 23 B channels + 1 D channel ISDN PRI is carrier over T1 and is used for PBX interconnect B PRI B B B D Mbps T1 and ISDN Training Confidential & Proprietary Information of VeEX Inc. 6

7 SIP Trunking SIP trunking: is a Voice over Internet Protocol (VoIP) service based on the signaling protocol Session Initiation Protocol (SIP) The SIP trunk connects customers equipped with SIP-based IP-PBX to the telephone network using IP protocol IP SIP Trunk Line IP PBX Extensions T1 and ISDN Training Confidential & Proprietary Information of VeEX Inc. 7

8 Why SIP Trunking? T1 lines and ISDN PRI leased lines are traditionally used by SMBs for PBX Voices Services Telcos T1 line are dedicated 4-wire lines, these lines are expensive and hard to maintain MSOs can use their existing HFC footprint to offer Business Voice Services at a lower cost. Cable operators benefit from new business revenue while SMBs benefit from decreased operating costs compared to leasing and maintaining T1 lines. Customer Premises installed with a DOCSIS modem and an Integrated Access Device (IAD), SMBs can keep their legacy ISDN PRI-PBX equipment. The IAD provides PRI to SIP translation. Customer Premises installed with a DOCSIS modem can use their existing SIP- PBX. VoIP Technology Confidential & Proprietary Information of VeEX Inc. 8

9 PRI Trunk MSO Network Customer Premises PSTN Internet Voice Gateway HFC Network POP - Firewall CMTS D3 Modem IAD T1 PBX SIP PRI Capacity up to 46 trunk lines on HFC network (2 x PRI) or more on Fiber network Use customer existing PRI PBX equipment IAD = Demarc converting PRI into SIP VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 9

10 SIP Trunk MSO Network Customer Premises PSTN Internet Voice Gateway HFC Network POP - Firewall CMTS D3 Modem SIP PBX SIP End to End SIP Capacity up to 46 trunk lines on HFC network (2 x PRI) or more on Fiber network Customer owned SIP PBX VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 10

11 VoIP Signaling SIP (Session Initiation Protocol) VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 11

12 Session Initiation Protocol - SIP Session Initiation Protocol (IETF RFC 3261) Signaling protocol used for the setup and signaling of VoIP calls SIP messages are exchanged between the SIP end points, which are the phones, and the SIP elements, which are the SIP servers, to establish the call SIP = Signaling SIP Server IP IP IP RTP= Media (Voice) VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 12

13 SIP Network Elements User Agent (UA): SIP endpoint. For example: an SIP phone or soft phones application running on PC Registrar: Server that receives the register request from users and keeps track of where to locate users Session Border Controller (SBC): Can be used in the network to provide services to the UAs connected to it, for security topology hiding or NAT traversal Gateways: Provide interconnect function between SIP and other networks (PSTN or H.323) VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 13

14 Simple Example of SIP Call INVITE OK Terminal A ACK Terminal B SIP requires only 3 messages to establish a call. INVITE: initiates the call 200 OK: response from called party ACK: confirmation VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 14

15 SIP URI All the SIP messages contain a TO and FROM fields, those fields are the SIP URI (Universal Resource Identifier) URI are in the format user@domain joe@domain.com joe@ POTS: @domain.com POTS: @ VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 15

16 SIP Call Flow Calling Party Registrar Proxy Called Party REGISTER 200 OK INVITE (with SDP) 100 Trying 180 Ringing 200 OK (with SDP) ACK INVITE (with SDP) 100 Trying 180 Ringing 200 OK (with SDP) ACK RTP Media = Voice BYE ACK BYE ACK ISDN Technology Training Confidential & Proprietary Information of VeEX Inc. 16

17 Speech Encoding VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 17

18 Speech Encoding Speech encoding is used to compress and encode human speech before it is transmitted through the network. It is used in IP and mobile telephony Voice Sampling 1 0 Analog Voice Signal Digitized Voice Signal Digital Signal transmission For the G.711 (PCM) codec, the voice is sampled at a rate of 8 khz (8,000 samples per second) and digitized using 8-bit samples before being transmitted VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 18

19 Speech Codecs G.711 G.711 Pulse Code Modulation (PCM), is a very commonly used waveform codec. Two different versions are available: G.711 µ-law used in North America, G.711 A-law used in the rest of the world Sampling frequency is 8kHz, bandwidth is 64 kbps Each voice packet contains 20ms speech Different encoding techniques achieve different results. There is usually a tradeoff between speech quality and bandwidth usage, computational delay and complexity. G.711 PCM encoding produces good quality speech but relatively poor bandwidth usage (64 kbps). It is generally seen as the toll-quality codec and has a MOS of 4.2 when no external impairments (e.g. delay, packet loss) are present. Other codecs such as G or G.729 achieve better bandwidth performance at the detriment of speech quality. Their MOS score is 3.8 and 3.9 respectively when no other impairments are present. VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 19

20 Speech Packetization Encoded speech is broken down into frames containing usually 20 ms or ms of speech. The amount of speech contained in a frame is described as p-time (packetization time). Speech frames are transported on the network using RTP (Real Time Transport Protocol) and UDP (User Datagram Protocol) IP Header UDP Header RTP Header Speech Codec (p-time varies depending on codec) UDP Header RTP Header Speech VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 20

21 IP Telephony Packets take different paths Address E Address A Voice is digitized and packetized At the receiving side, voice packets are reassembled, reordered, and played out Address D Address B Address C Voice is digitized, packetized and sent over the IP network. As voice travels through the network, packets may be impaired as a result of network impairments (i.e., jitter, delay, Packet loss) VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 21

22 VoIP Impairments Confidential & Proprietary Information of VeEX Inc. 22

23 VoIP over HFC Voice trunking requires more HFC bandwidth than the traditional residential services Up to 46 simultaneous trunk lines can be in use Careful traffic engineering at the CMTS and in the backbone is required Confidential & Proprietary Information of VeEX Inc. 23

24 Common VoIP Impairments Network delay High levels of delay (generally over 200 milliseconds round trip) can cause problems with conversational interaction. Delay can also make echo problems more obvious and annoying The sources of delay on a VoIP include codec encoding/decoding delay, packetization time, network transmission delay VoIP Technology Confidential & Proprietary Information of VeEX Inc. 24

25 Common VoIP Impairments Jitter Jitter is the variation in packet transit delay it is cause by queuing, congestion, network route changes Although some level of jitter is expected and taken care of in the jitter buffer, excessive jitter will cause packet discard, degrading speech quality. VoIP Technology Confidential & Proprietary Information of VeEX Inc. 25

26 Common VoIP Impairments Packet Loss Occurs due to a variety of reasons including Link failure, high levels of congestion, router buffer overflow, physical links problems Most codecs are equipment with Packet Loss Concealment algorithm that mask the effects of lost packets. These algorithms are inefficient if packet loss comes in bursts and bursty packet loss has a severe impact on voice quality even if the average packet loss rate for the call is low. Out of order or Duplicate packets Occurs due to a variety of reasons including Link failure, high levels of congestion, router buffer overflow, physical links problems Out of order or duplicate packets are taken care of by the jitter buffer, but excessive count could lead to packet discard and indicate network issues VoIP Technology Confidential & Proprietary Information of VeEX Inc. 26

27 Echo in VoIP Networks (1) Echo Sources of Echo: Reflection on the 2/4 Wire interface (Hybrid echo) Acoustic Echo created by Created by the voice reflected back from the microphone to the speaker due to device issues or reflective environment Echo reported by Echo Return Loss (ERL) 55 db ERL represents a low echo 15 db ERL represents a high echo Acoustic Isolation Echo Ambient Acoustic Coupling Requires >45 db isolation Poor handset or headset design VoIP Technology Confidential & Proprietary Information of VeEX Inc. 27

28 How is Voice Quality Measured? Confidential & Proprietary Information of VeEX Inc. 28

29 Subjective Voice Quality Measurement Subjective Test: MOS (Mean Opinion Score): panel of listeners rate the call quality Source Channel Rating Speech Quality 5 Excellent 4 Good 3 Fair 2 Poor 1 Unsatisfactory VoIP Technology Confidential & Proprietary Information of VeEX Inc. 29

30 Objective Voice Quality Measurement Objective Tests (machine tests): ITU-T P.862 (PESQ) determine the distortion introduced by a transmission system or codec by comparing an original reference file sent into the system with the impaired signal that came out VoIP Technology Confidential & Proprietary Information of VeEX Inc. 30

31 Testing Topology PRI Voice Trunk Testing PSTN Voice Gateway Internet POP - Firewall CMTS D3 Modem IAD T1 PBX Bi-directional PESQ Voice Quality Testing D3 Modem SIP PBX SIP Trunk Testing VoIP Technology Training Confidential & Proprietary Information of VeEX Inc. 31

32 Examples of VoIP Testing VoIP Technology Confidential & Proprietary Information of VeEX Inc. 32

33 Thank you. Any questions? Tel: Confidential & Proprietary Information of VeEX Inc. 33