TCS6 Chapter 1 : VoIP Fundamentals IRISET
Welcome message from author
This document is posted to help you gain knowledge. Please leave a comment to let me know what you think about it! Share it to your friends and learn new things together.
Transcript
TCS6Chapter 1 : VoIP Fundamentals
IRIS
ET
VOIP
• VoIP means "Voice over Internet Protocol" is a revolutionarytechnology that allows us to make telephone calls and send faxes overIP based data networks with a suitable quality of service and superiorcost/benefit a part from our regular phone line.
IRIS
ET
ADVANTAGES• Reduced cost of Communication
• Flexibility
• Accessibility
• Portability
• Redundancy or High Availability
• Ease of deployment
• Simplification of transport networks
• Value added services
• Scalability
IRIS
ET
DISADVANTAGES
• Reliability
• Power Outage
• Security
• Processor Issues
IRIS
ET
Elements of VoIP
IRIS
ET
IP-PBX (server)
• This is a common server on which VoIP software is installed andconfigured to provide IP- Telephony. Some vendors sell proprietyservers (which include Hardware and VoIP Software) as well as in thename of features.
• It is a PBX (Private branch Exchange) phone system that uses IP(Internet Protocol) data networks to manage call switching betweenclients, connecting multiple protocols, route calls and handle othermessaging services. It includes advanced communication features,like voicemail to email, etc.IR
ISET
VoIP SOFTWARE• VoIP software enables telephone-like voice conversations across IP based
networks, which is to be installed upon IP-PBX (server). Additional softwareapplications include conferencing servers, intercom systems, virtual foreignexchange services (FXOs) and adapted telephony software whichconcurrently support VoIP and public switched telephone network (PSTN)like Interactive Voice Response (IVR) systems, dial in dictation, on hold andcall recording server, etc. can be installed on IP-PBX(Server)
• These are categorized as two types
Open sources IP-Telephony software’s• Ex: Asterisk, Free Switch, Free PBX, OpenSIPS, YATE, Issabel, Kamailio, Etc.
Proprietary IP-Telephony Software’s• Ex: Cisco, 3CX Phone system, Astra, Alcatel-Lucent, Avaya Application Server 5300,NEC, Nortel,
Tadiran Telecom, Unify Openscape, Nokia Siemens networks, etc.
IRIS
ET
IP TELEPHONES• Looks like the ordinary traditional phone.
• Works like a modern mobile phone.
• Dial tone is local. Dial the number and then send the same to theexchange.
IRIS
ET
VoIP GATEWAYS• VoIP Gateways: VoIP gateway is a device that converts telephony traffic
into an IP for transmission over a data (IP) network and vice-verse. They can be Analog or Digital gateways.
• Analog VoIP gateways are used to connect ordinary Push-Button telephone to a VoIP system.
Analog gateways – FXS, FXO• FXS used to connect the PBT (station) to the gateway.
• FXO used to connect CO Line (office) to the gateway.
Digital gateways – PRI gateways• PRI GW used to connect a PRI line to the VoIP system.IR
ISET
Network Switches
• These are normal switches but support Power over Ethernet (PoE) on each port.
• IEEE 802.3af-2003
• IEEE 802.3at-2009
30W of DC power/Power available – 25.5W
IRIS
ET
Dialing options• ATA Analog Telephone Adaptor/FXS Gateways:
It use a standard Analog Push Button phone to make VoIP calls. TheATA is an analog to digital converter. It takes the analog signal from yourtraditional phone and converts it into digital data for transmission over theInternet.
• IP Phones:
It look like normal phones with a handset, cradle and buttons. Butinstead of RJ11, IP phones connected through RJ45 Ethernet connector.
• Soft Phones:
A small software applet known as soft phone installed on to a PC orSmart Phone can be used to make and receive calls. With headset one cancommunicate.
IRIS
ET
VoIP Issues
• Voice signals in VoIP calls are encoded into digital data, which are compressed(reducing the number of bits per sample without losing the quality of thespeech) so as to make it less bulky for transmission over the Data Network. Itis called as Speech Coding.
• Advantages:
1. Efficient use of Bandwidth: Compress to lower bit rate per user, makes moreusers can accommodate within the given bandwidth.
2. Hardware complexity: Computation requirement and power consumptionreduces.
3. Security: improved security can be maintained.
4. Maintenance: Operations and Maintenance cost reduces.IR
ISET
Speech Coding techniques
Waveform coders:
• Waveform coders digitize the speech signal on a sample-by-samplebasis. Its main goal is to make the output waveform to resemble theinput waveform. So waveform coders retain good quality speech.Waveform coders are low complexity coders, which produce highquality speech at data rates above and around 16 Kbps.
• Time domain Ex. PCM, DPCM and ADPCM and
• Frequency domain Ex: Sub-band Coders and Adaptive transformCoders IR
ISET
Speech Coding techniques
Source coders (vocoders):
• The encoder builds a set of parameters from voice, derives theperceptual feature of the voice and sends the parameters to thereceiver. The receiver has a synthesizer and reproduces the originalvoice based on the parameters received.
• The reproduced voice sounds "synthetic” and is not good enough fortelephony. Very low bit rates (1 to 4 kbps) with reasonable quality andworks in the frequency domain.
• Ex: LPC (Linear Predictive Coders) and MELP (Mixed Excitation Linearpredictive).
IRIS
ET
Speech Coding techniquesHybrid coders:
• To fill the gap between waveform and source codecs. Hybrid codersoperate at medium bit-rates between those of waveform codersandSource coders and produce high quality speech than source coders.
• Waveform coders are capable of providing good quality speech at bitrates up to 16 Kbps, but are of limited use at rates below this.
• Source coders on the other hand can provide intelligible speech at 2.4kilobits per second and below, but cannot provide natural soundingspeech at any bit rate.
• The most successful and commonly used are time domain Hybridcoders are Analysis-by-Synthesis (AbS) codecs. MPE (Multi-pulseExcited), RPE (Regular pulse Excited) and CELP (Code-excited linearpredictive).
IRIS
ET
Voice coding Standards
• G.711 - Also known as Pulse Code Modulation (PCM), it is the ITU-Tinternational standard for encoding telephone audio on a 64 kbpschannel. PCM samples the signal 8000 times a second; each sample isrepresented by 8 bits for a total of 64 kbps.
• G.721- An ITU-T standard codec that uses Adaptive Differential PulseCode Modulation (ADPCM); a form of Pulse Code Modulation (PCM),to produce a digital signal with a lower bit rate than standard PCM.This ITU standard for speech codecs uses ADPCM on a 32 kbpschannel. G.721 was first introduced in 1984IR
ISET
Voice coding Standards
G.722 : An ITU-T standard codec that uses sub-band adaptive differentialpulse code modulation (SB-ADPCM) within a bit rate of 64 kbps. Thesystem is referred to as 64 Kbps (7 kHz) audio coding. SB-ADPCM splits thefrequency band into two sub-bands(higher and lower) and the signals ineach sub-band are encoded using ADPCM.
G.722.1 - An ITU-T standard codec describes a low complexity encoder anddecoder that may be used for 7 kHz bandwidth audio signals working at 24kbps or 32 kbps.
G.722.2 - An ITU-T standard codec describes the high quality Adaptive Multi-Rate Wideband (AMR-WB) encoder and decoder that is primarily intendedfor 7 kHz bandwidth speech signals. AMR-WB operates at a multitude of bitrates ranging from 6.6 kbps to 23.85 kbps.
IRIS
ET
Voice coding StandardsG.723 - An ITU-T standard codec that uses Adaptive Differential Pulse Code
Modulation (ADPCM) standard for speech codecs on a 24 and 40 kbpschannel.
G.723.1 - An ITU-T standard codec uses dual rate speech coder used for POTS,Video Conferencing and Telephony. The standard supports 6.3 Kbps data ratebased on Multiple-pulse maximum likelihood quantization (MP-MLQ) with anaudio quality along with MOS score of 3.98 and 5.3 kbps data rate based onAlgebraic code Excited linear predictive (ACELP).
G.726 - An ITU-T Adaptive Differential Pulse Code Modulation (ADPCM)standard speech codec used for the transmission of voice at rates on 16, 24,32, and 40 kbps channels. G.726 supersedes both G.721 and G. 723 as itincludes both of these standards plus includes the new standard for the 16kbps rate. G.726 was the standard codec used in Digital Enhanced CordlessTelecommunications (DECT) wireless phone systems.
IRIS
ET
Voice coding Standards
G.727 - A specialized version of the ITU-T G.726 protocol that isintended for packet based systems using the Packetized VoiceProtocol (PVP). G.727 uses 5, 4, 3 and 2- bit/sample embeddedadaptive Differential Pulse Code Modulation (ADPCM).
G.728 - An ITU-T speech coding standard that uses Low Delay CodeExcited Linear Prediction (LD-CELP) operating at 16 kbps compressionat a sampling rate of 8,000 samples per second. The algorithmiccoding delay of G.728 is 0.625ms. G.728, when compared to G.726delivers close to the same voice quality but uses only one-half thebandwidth. IR
ISET
Voice coding Standards
G.729 - An ITU-T audio data compression standard that operates at 8 kbpsusing a conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP). This algorithm for voice compresses voice audio in 10 millisecondframes. G.729 is commonly used in in Voice over IP (VoIP) applicationsbecause of its inherently low bandwidth requirement.
• Extensions to the G.729 standard include the following;
G.729a (G.729 Annex A) - Compatible with G.729 Annex A specifies a coderwith several simplifications, including code book search routines. Thesemodifications are known to often result in a slightly lower voice quality.
G.729b (G.729 Annex B) - Compatible with G.729, Annex B specifies a coderthat uses Discontinuous Transmission (DTX), Voice Activity Detection (VAD),and Comfort Noise Generation (CNG) to reduce bandwidth usage.Bandwidth is reduced bypreventing the transmission of non-voice duringperiods of silence (silence detection).
IRIS
ET
Voice coding Standards
GSM-06.10 – The bit rate of the codec is 13 kbps, it uses Linear predictiveCoding with Regular pulse excitation (LPC_RPE codec) suitable for GSM.
iLBC ( Internet Low Bit rate Codec) – A free speech codec suitable for robustvoice communication over IP. The codec is designed for narrow bandspeech and results in a payload bit rate of 13.33 kbps with an encodingframe length of 30 ms and 15.20 kbps with an encoding length of 20 ms itis being used by many pc-pc applications Google talk, Yahoo messengerMSN messenger.
Speex- Speex is an Open Source/Free Software patent-free audiocompression format designed for speech. Licensed under the Xiph.orgvariant of the BSD license. It is a Variable Bitrate (VBR) codec, which meansthat it is able to dynamically modify its bitrate (2 to 44 kbps) to respond tochanging network conditions. Asterisk, Linphone and Ekiga uses speexcodecs
IRIS
ET
Voice coding Standards
Silk - Skype Limited announced that SILK can use a sampling frequencyof 8, 12, 16 or 24 kHz and a bit rate from 6 to 40 kbps. It can also usea low algorithmic delay of 25ms (20ms frame size + 5ms look-ahead).The SILK binary SDK is available. The SILK codec is patented andlicensed separately from the SILK SDK. The codec is open-source,freeware, available royalty free with restrictions on use anddistribution.
Opus - Opus is a totally open, royalty-free, highly versatile audio codec.Opus is unmatched for interactive speech and music transmissionover the Internet, but is also intended for storage and streamingapplications. It is standardized by the Internet Engineering Task Force(IETF) as RFC 6716 which incorporated technology from Skype’s SILKcodec and Xiph. Org’s CELT codec.
IRIS
ET
Delay / Latency• VoIP Delay or Latency is characterized as the amount of time it takes
for the speech to exit the speaker's mouth and reach the listener'sear.
IRIS
ET
Delay / LatencyCoder delay: Coder delay depends on the used codec. It has two
components: the frame size delay and the look-ahead delay. Theirvalues are exactly defined for any particular coder.
Packetization delay: The packetization delay rises during the process ofdata blocks encapsulation into packets, which are consequentlytransmitted by the network.
Serialization delay: Serialization delay depends on the transmissionrate of the used interface. The transmission of packets takessometime which depends on the transmission medium rateand onthe size of packet. IR
ISET
Delay / LatencyPropagation delay: This delay relates to the physical environment of the
propagation medium. It depends on the transmission technology used, inparticular on the distance over which the signal is transmitted.
Queuing delay: This delay occurs in active elements of the transmissionnetwork, in particular in the router queues. When packets are held in aqueue because of congestion on an outbound interface, the result isqueuing delay. This delay is the most significant part of the jitter. It is avariable delay.
De-jitter delay: Speech is a constant bit-rate service, the jitter from all thevariable delays must be removed before the signal leaves the network. Thisis accomplished by a dejitter buffer at the far-end (receiving)router/gateway. The de-jitter buffers can be adaptive, but the maximumdelay is fixed.
IRIS
ET
Delay / Latency
De-packetization delay: The de-packetization is a reverse packetizationand therefore the size of de-packetization delay of one block in the frame is in correlation with its packetization delay.
De-compression delay: The decompression delay depends on the compressing algorithm selection. On an average, the decompression delay is approximately 10 % of the compressing codec delay for each voice block in the packet. IR
ISET
Delay / Latency• As per ITU-T – G.114 Recommendation "It specifies that for good voice quality,
one-way end-to-end (i.e., "mouth-to-ear" in the case of speech) delays of lessthan 150 ms, if delays can be kept below this figure, most applications, bothspeech and non-speech, will experience essentially transparent interactivity.
• While delays above 400 ms are unacceptable for general network planningpurposes, it is recognized that in some exceptional cases this limit will beexceeded. An example of such an exception is an unavoidable double satellitehop for a hard-to-reach location.
IRIS
ET
Jitter• Jitter is a measure of the variability of delay
• It is the average variation in the delivery time between packets.
• If the variation is greater than 20 milliseconds (>= 30 ms as per cisco),jitter creates audible voice-quality problems similar to those createdby high latency.
IRIS
ET
ECHO• When users of VoIP make calls they could hear their own voice
reflected to their phones speaker after a few milliseconds. Thisannoying phenomenon is known as Echo.
IRIS
ET
Causes and Elimination of ECHO• Hybrid echo: An improperly balanced hybrid circuit, such as a two-wire to
four-wire interface, allows the transmit signal to appear on the receivepath.
• Acoustic echo: Poor design in telephone handsets and hands-free devicesallows the microphone to pick up sounds from the earpiece
Ways of dealing with echo problems:
• Elimination at source: Ensure that all of the hybrids are balanced.
• Echo suppression: An echo suppressor is a simple voice-activated switchwhich turns off transmission from talker to listener whenever the talker issilent.
• Echo Cancellation: Echo cancellation uses a mathematical approach tosubtract exactly the right portion of the transmitted signal from thereturn signal to eliminate the echo.
IRIS
ET
Packet Loss• Voice is treated as normal data in IP Networks. Due to this fact, the voice
packets are vulnerable to the unfortunate cases of being dropped whenthe traffic is high and the network is congested.
• Re-transmission of lost data packets can solve the problem in datatransmission, but voice requires real-time transmission.
• Packet loss can be compensated in the end point by using algorithms likePacket Loss Concealment (PLC) or Packet Loss Recovery (PLR).
• Network device should be able identify the VoIP packets. The process ofpacket identification is termed as classification and it is the basicfoundation towards achieving a desired QoS.
• Resource Reservation Protocol (RSVP) is another mechanism whichprovides QOS.
IRIS
ET
Voice Activity Detection• VoIP is voice activity detection (VAD) is enabled. VAD works by detecting the
magnitude of speech in decibels (dB) and deciding when to cut off the voice frombeing framed.
• It waits a fixed amount of time before it stops putting speech frames in packets. Thisfixed amount of time is known as hangover and is typically 200 ms.
• VAD is unable to distinguish between speech and background noise. This also isknown as the signal-to-noise threshold. VAD disables itself at the beginning of the call.
• Inherent problem with VAD is detecting when speech begins. Typically the beginningof a sentence is cut off or clipped. This phenomenon is known as front-end speechclipping. Usually, the person listening to the speech does not notice front-end speechclipping.
IRIS
ET
QOS1. 802.1p/Q (layer 2 QoS)
• Partitioning of a LAN into separated domains (usage of 12-bit 802.1QVLAN ID).
• Priority field (3-bit 802.1p p-tag) for QoS.
IRIS
ET
QOS2. QoS at layer 2.5: MPLS Multi-Protocol Label Switching
• MPLS switches IP traffic flows on layer 2 thus improving networkperformance (more throughput).
• MPLS combines IP routing (addressing) and fast forwarding of traffic(layer 2 switching).
• MPLS LSPs (Label Switched Paths) can be assigned certain QoS (like ATMPVCs).
IRIS
ET
QOS3. QoS at layer 3: TOS Type of Service = DiffServ (Differentiated Services)
• DiffServ contains 2 main components:
• 1. Classification/prioritization of packets in forwarding path based onDSCP IP header field.
• 2. Policy and allocation of priorities along the transmission path.
IRIS
ET
QOS3. QoS at layer 3: TOS Type of Service = DiffServ (Differentiated Services)
• TOS field was too un-flexible and redesigned to a single field named DSCP.DSCP contains a number that indicates the PHB to be applied to the IPpacket.
• Packets are classified (and DSCP field marked) at the ingress into adomain (e.g. AS Autonomous System). Intermediate routers in domain Bprioritize packets according to the DSCP field in IP header. Domain Begress router shapes and schedules packets.as shown in below figureIR
ISET
QOS4. QoS at layer 3: RSVP (IntServ Integrated Services)
• RSVP (Resource ReSerVation Protocol) is an end-to-end protocol forbandwidth and latency requirements allocation and reservation.
• RSVP uses standard IP routing protocols for deciding where to allocateresources. Since RSVP uses receiver-based allocation (as opposed tosender-based allocation) multicast can be easily supported (reservationsflow towards the root of the multicast tree).
IRIS
ET
QOS5. Queueing strategies
1. FIFO-First In First Out:
2. Priority queueing PQ (SP –Strict Priority Queue):
3. Round Robin RR:
4. Class Based queueing CB:
5. Weighted Fair Queueing (WFQ):
IRIS
ET
QOS6. Active Queue Management –AQM
A. Random Early Detection-RED/ Weighted RED (WRED):
B. Explicit Congestion Notification – ECN
C. CoDel (Controlled Delay):
IRIS
ET
QoS Requirements for Voice
• Voice calls, either one-to-one or on a conference connection capability, require the following:
• ≤ 150 ms of one-way latency from mouth to ear (per the ITU G.114 standard)
• ≤ 30 ms jitter
• ≤ 1 percent packet loss
• 17 to 106 kbps of guaranteed priority bandwidth per call (depending on the sampling rate, codec, and Layer 2 overhead)
• 150 bps (plus Layer 2 overhead) per phone of guaranteed bandwidth for voice control traffic
IRIS
ET
CHAPTER - 2VoIP
SIGNALLING & MEDIA TRANSMISSION PROTOCOLSIR
ISET
VOIP Signaling and Media Transmission Protocols
• ITU: International Telecommunication Union
H.323 -ITU recommends for “ Packet based Multimedia communication systems”.
- Most common VoIP protocol
- Distributed Architecture
• IETF: Internet Engineering Task Force
SIP: Session Initiation Protocol
-IETF RFC 2543 ,3265
-Distributed Architecture
Real-time Transport Protocol – RTP
- A transport protocol for real-time application
- IETF RFC 1889 ,3550
- Provides transport for audio/media of VoIP communication
-Used by All of VoIP signaling protocols
IRIS
ET
VOIP PROTOCOL STANDARDS
• MGCP Media Gateway Control Protocol
- IETF RFC 2075 ,3435
- Centralized Architecture for Multimedia applications such as VoIP
• H.248 Gateway Control Protocol
Collaboration between ITU & IETF referred to as IETF RFC 2885,3015 (MEGACO) IR
ISET
Centralized Architecture • Mostly used in older networks
• Worked well for basic telephony services
• Trade off between easy management and endpoint innovation
• Associated with MGCP and H.248
• Intelligence focused in centralized Gateway unit (media agent)
• Endpoints are relatively or completely dumb.
IRIS
ET
Distributed Architecture • Associated with H.323 and SIP protocol
• Allows Network Intelligence to be distributed between endpoints and control devices.
• Intelligence:
Call state
Calling features
Calling routines
any aspect of call handling
• More flexible
• VoIP is treated like any other distributed IP application
• Well understood by engineers who design and run IP data networks
• More complex than the Centralized Architecture
IRIS
ET
VOIP PROTOCOLS
IRIS
ET
H.323 Network Architecture and Components
IRIS
ET
H.323 ComponentsH.323 terminal
• Terminals or a client, is an end point where H.323 data streams andsignalling originate and terminate which includes Video I/O equipment (IPVIDEO Phone), Audio I/O equipment (IP Phone), User Data Applicationsand System Control User Interfaces(running on PC).
• Terminals can be IP-Telephone, IP-Video phones, IVR Devices, Voicemail Systems, Softphones
Gateways:
• The Gateways is composed of a "Media Gateway Controller"(MGC) and a"Media Gateway" (MG), which may co-exist or exist separately. Gatewaysconnect H.323 networks to other networks, including the PSTN, ISDN,H.320 systems, other H.323 networks (proxy), etc.
IRIS
ET
H.323 ComponentsGatekeeper:
1. Address Translation: Provides endpoint IP addresses from H.323 aliases (978-555-4567 204.124.46.19) E.164 Number Network address ----- ENUM
2. Admissions Control: Provides authorized access to H.323 using the Admission Request/Admission Confirm/Admission Reject (ARQ/ACF/ARJ) messages.
3. Bandwidth Control: consists of managing endpoint bandwidth requirements using Bandwidth Request/Bandwidth Confirm/Bandwidth Reject (BRQ/BCF/BRJ) messages.
4. Zone Management: Provided for registered terminals, gateways, and MCUs IRIS
ET
H.323 ComponentsMulti Conference Units (MCUs):
• Centralized multipoint conference:
• Decentralized multipoint conference:
• Hybrid multipoint conference:
IRIS
ET
H.323 Components And Protocols
IRIS
ET
H.323 Protocol Suite
IRIS
ET
Call Exchange (direct mode)
IRIS
ET
IRIS
ET
H.323 Call-Signaling ProcessThere are five general steps in the H.323 signaling process: setup/teardown, capabilities negotiation, open media channel, perform call, and release.
Setup/Teardown
To initiate an H.323 call, H.225 is required for the setup process.
The following are the most commonly used signaling messages :
Setup: A forward message sent by a calling entity in an attempt to establish a connection with the called entity
Proceeding: A backward message sent from the called entity to the calling entity to inform that call establishment procedures were initiated
IRIS
ET
H.323 Call-Signaling Process
Alerting: A backward message sent from the called entity to inform that called party ringing was initiated
Connect: A backward message sent from the called entity to the calling entity that the called party answered the call. The connect message can contain the transport UDP/IP address for H.245 control signaling
Release: sent by endpoint initiating disconnect
IRIS
ET
SIP• Proposed Standard described in IETF RFC 2543, 3261
• ASCII based and Application-layer control protocol
• A signaling protocol for initiating, managing and terminating voice and audio session across packet networks with one or more participants .
• Text-based protocol with highly extensible
• Session can be
1. Call between two simple telephone
2. Collaborative multi-media conference session, etc.IRIS
ET
SIP Functionality• User location
• User availability
• User Capabilities
• Session setup
• Session Management
• Session termination
IRIS
ET
FUNCTIONALITIES• SIP serves 4 major functionalities
1. It allows to locate the user ( i.e translating user’s name to their current network address)
2. Inviting the user for session
-negotiation so that all of the participants in a session can agree on the features to be supported among them
3. Delivering the session description
- call management such as adding ,dropping or transferring participants.
4. Terminate the session
IRIS
ET
SIP entitiesClient: Network element that sends SIP requests and receives SIP responses.
• User Agent (UA)• User agent client (UAC)• User agent server (UAS)
Server: Network element that receives requests in order to service them and sends back responses to those requests.
• Proxy server• Stateless proxy server• Stateful proxy server
• Redirect server
• Registrar server
IRIS
ET
Client
• User Agent Client (UAC)
–Application which originates SIP requests
• User Agent Server (UAS)
–Application which contacts user upon receiving SIP request, and–Returns user’s response on his behalf
- Accepts, rejects or redirects
• User Agent (UA)
–Application which contains both UAC & UAS and exchange request/response messages
UA is a piece of software that can be placed in a computer or a laptop
Therefore, SIP can offer –Various telephony services,
e.g., ▪Internet phones-to-Internet phones▪Internet phones-to-PSTN phones▪PC phones-to-PC phones
IRIS
ET
SIP SERVERS
• proxy server: The Proxy Servers are application layer routers that forward SIP request & responses
• Redirect server: A redirect server is a server that accepts a SIP Requests & then return the location of another SIP user agent & server where the user might be found.
• Registrar server: A registrar is a server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and offer location services
• Location Server: the addresses registered to a Registrar are store in a Location server
IRIS
ET
SIP NETWORK ELEMENTS
IRIS
ET
SIP ARCHITECTURE
IRIS
ET
SIP ARCHITECTURE
IRIS
ET
IRIS
ET
SIP operation
USER
AGENT 1
USER
AGENT2
1.invitation 2.invitation
3. OK4. OK
5.Acknowledge 6.Acknowledge
7. Audio/Video data 7. Audio/Video data
From: Thomas Doumas
Next Generation Telephony: A Look at Session Initiation Protocol White Paper
IRIS
ET
SIP Protocol Structure
From: http://docs.sun.com/app/docs/doc/821-0203/6nl988v7d
▪Client
▪Sending Request
▪Receiving Response
▪Server
▪Receiving Request
▪Sending Response
▪Framing
▪Error Handling
▪Client
▪INVITE Transaction
▪ACK
▪Non INVITE Transaction
▪Matching Requests to
Client Transactions
▪Server
▪INVITE Transaction
▪Non INVITE Transaction
▪Matching Requests to
Server Transactions
▪Error Handling
IRIS
ET
SIP Protocol Suite
IRIS
ET
Other IETF ProtocolsSIP agents or applications need other protocols for the following:
• To describe the characteristics of a session:
• To handle media:
• To support functions:
Protocols
• DNS = Domain Name System
• RSVP = Resource Reservation Protocol
• SDP = Session Description Protocol
• TLS = Transport Layer Security
• STUN = Simple traversal of UDP through Network address
IRIS
ET
SIP AddressingSIP addresses are typically referred to as SIP URI.• A SIP URI is typically and e-mail-type address with a form such as one of the
given following• sip: user@domain:port• sip: user @host:portThe user field identifies a user by name, such as "amruth", by telephonenumber, such as "04027785900" or by MAC address of the system, such as"AA346823BB45", within the context of a domain or a host. The port is anoptional field. If no port is specified, the default port for a SIP URI is 5060.• Examples of SIP URIs are as follows1. sip: [email protected]. sip: [email protected]. sip: [email protected]• The public SIP address of a user or a resource is referred to as an Address-of-
Record (AOR)
IRIS
ET
SIP REQUESTS•INVITE
–Request initiation of a session
Most common and important
•ACK
–Confirm that a session has been initiated
•BYE
–Request termination of a session
•OPTIONS
–Query a host about its capabilities
•CANCEL
–Cancel a pending request
•REGISTER
–Inform a redirection server about the user’s current location
IRIS
ET
SIP REQUESTS•SUBSCRIBE
– To acquire updated information on whether a User agent is online, busy, offline, and so on.
•NOTIFY
– To sends presence information on whether a User agent is online, busy, offline, and so on.
•PUBLISH
–PUBLISH is most useful when there are multiple sources of event information, such as a number of devices sharing the same AOR.
•REFER
–When the URI is a sip or sips URI, the REFER is probably being used to implement a call transfer service.
•MESSAGE
–used to transport instant messages (IM) using SIP
•PRACK
–used to acknowledge receipt of reliably transported provisional responses (1xx)
•UPDATE
–used to modify the state of a session without changing the state of the dialog.
IRIS
ET
SIP RESPONSES•1xx-Provisional (Informational)
–Request received, continued to process request, e.g., 180 = Ringing
•2xx–Success
–Action was successfully received, understood, and accepted, e.g., 200 = OK
•3xx –Redirection
–Further action must be taken to complete the request, e.g., 305 = Use Proxy
•4xx-Client Error
–The request contains bad syntax or cannot be fulfilled at the server, e.g., 484 = Address Incomplete
•5xx -Server Error
–The server failed to fulfill an apparently valid request, e.g., 500 = Internal Server error
•6xx-Global Failure
–The request is invalid on any server, e.g., 600 = Busy
IRIS
ET
Request Message example
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP alice_ws.radvision.com
From: Alice A. <sip:[email protected]>
To: Bob B. <sip:[email protected]>
Call-ID: 2388990012@alice_ws.radvision.com
CSeq: 1 INVITE
Subject: Lunch today.
Content-Type: application/SDP
Content-Length: 182
{BODY}
v=0 o=Alice 53655765 2353687637 IN IP4 128.3.4.5 s=Call from Alice. c=IN IP4 alice_ws.radvision.com M=audio 3456 RTP/AVP 0 3 4 5
IRIS
ET
Response Message example
SIP/2.0 200 OK
Via: SIP/2.0/UDP alice_ws.radvision.com
From: Alice A. <sip:[email protected]>
To: Bob B. <sip:[email protected]>;tag=17462311
Call-ID: 2388990012@alice_ws.radvision.com
CSeq: 1 INVITE
Content-Type: application/SDP
Content-Length: 200
{BODY}
v=0 o=Bob 4858949 4858949 IN IP4 192.1.2.3 s=Lunch c=IN IP4 machine1.acme.com m=audio 5004 RTP/AVP 0 3
IRIS
ET
Session establishment and termination
UAC UAS
1: INVITE [email protected]
2. 100 Trying
3. 180 Ringing
4. 182 Queued, 2callers ahead
5. 182 Queued, 1callers ahead
6. 200 OK
7.ACK
1: BYE [email protected]
2. 200 OK
IRIS
ET
Registration
Bob’s
softphone
Biloxi.com
registrar
Register
200 OKREGISTER sip:registrar.biloxi.com SIP/2.0Via: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=z9hG4bKnashds7Max-Forwards: 70To: Bob <sip:[email protected]>From: Bob <sip:[email protected]>;tag=456248Call-ID: 843817637684230@998sdasdh09CSeq: 1826 REGISTERContact: <sip:[email protected]>Expires: 7200Content-Length: 0
SIP/2.0 200 OKVia: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=z9hG4bKnashds7
;received=192.0.2.4To: Bob <sip:[email protected]>;tag=2493k59kdFrom: Bob <sip:[email protected]>;tag=456248Call-ID: 843817637684230@998sdasdh09CSeq: 1826 REGISTERContact: <sip:[email protected]>Expires: 7200Content-Length: 0
IRIS
ET
Call Redirection
UAC Redirect server
sip.acme.com
Location
Service
UASgw.telco.com
1. INVITE [email protected]
4. 302 Moved [email protected]
5.ACK
6. INVITE [email protected]
7. 200 OK
8.ACK
IRIS
ET
Call ProxyBob’s
softphoneBiloxi.com
Proxy
Atlanta.com
ProxyAlice’s
softphone
1.INVITE
3..INVITE
5..INVITE2.100 Trying
4.100 Trying
6.180 Ringing
1. INVITE Alice -> atlanta.com proxy
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8Max-Forwards: 70To: Bob <sip:[email protected]>From: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 142
2. 100 Trying atlanta.com proxy -> Alice
SIP/2.0 100 TryingVia: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>From: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContent-Length: 0
3 INVITE atlanta.com proxy -> biloxi.com proxy
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1Max-Forwards: 69To: Bob <sip:[email protected]>From: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 142
F4 100 Trying biloxi.com proxy -> atlanta.com proxy
SIP/2.0 100 TryingVia: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>From: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContent-Length: 0
5.INVITE biloxi.com proxy -> BobINVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1Max-Forwards: 68To: Bob <sip:[email protected]>From: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 142
6.180 Ringing Bob -> biloxi.com proxySIP/2.0 180 RingingVia: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1;received=192.0.2.3Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710Contact: <sip:[email protected]>CSeq: 314159 INVITEContent-Length: 0
IRIS
ET
Bob’s
softphoneBiloxi.com
Proxy
Atlanta.com
ProxyAlice’s
softphone
1.INVITE
3..INVITE
5..INVITE2.100 Trying
4.100 Trying
6.180 Ringing
7.180 Ringing
8.180 Ringing9. 200 OK
10. 200 OK
11. 200 OK
12. ACK
7.180 Ringing biloxi.com proxy -> atlanta.com proxy
SIP/2.0 180 RingingVia: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710Contact: <sip:[email protected]>CSeq: 314159 INVITEContent-Length: 0
8 180 Ringing atlanta.com proxy -> Alice
SIP/2.0 180 RingingVia: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710Contact: <sip:[email protected]>CSeq: 314159 INVITEContent-Length: 0
9 200 OK Bob -> biloxi.com proxySIP/2.0 200 OKVia: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1;received=192.0.2.3Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 131
10. 200 OK biloxi.com proxy -> atlanta.com proxySIP/2.0 200 OKVia: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 131
11 200 OK atlanta.com proxy -> Alice
SIP/2.0 200 OKVia: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8;received=192.0.2.1To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 131
12 ACK Alice -> Bob
ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds9Max-Forwards: 70To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 314159 ACKContent-Length: 0 IR
ISET
Bob’s
softphoneBiloxi.com
Proxy
Atlanta.com
ProxyAlice’s
softphone
1.INVITE
3..INVITE
5..INVITE2.100 Trying
4.100 Trying
6.180 Ringing
7.180 Ringing
8.180 Ringing9. 200 OK
10. 200 OK
11. 200 OK
12. ACK
Media Session
13.BYE
14. 200 OK
13 BYE Bob -> Alice
BYE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10Max-Forwards: 70From: Bob <sip:[email protected]>;tag=a6c85cfTo: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 231 BYEContent-Length: 0
14 200 OK Alice -> Bob
SIP/2.0 200 OKVia: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10From: Bob <sip:[email protected]>;tag=a6c85cfTo: Alice <sip:[email protected]>;tag=1928301774Call-ID: a84b4c76e66710CSeq: 231 BYEContent-Length: 0
IRIS
ET
SIP Protocol StructureBob’s
softphoneBiloxi.com
Proxy
Atlanta.com
ProxyAlice’s
softphone
1.INVITE
3..INVITE
5..INVITE2.100 Trying
4.100 Trying
6.180 Ringing
7.180 Ringing
8.180 Ringing9. 200 OK
10. 200 OK
11. 200 OK IRIS
ET
IRIS
ET
IRIS
ET
MGCP- Media Gateway Control Protocol
• The Media Gateway Control Protocol (MGCP) also comes to us from the IETF.
• It is a Protocol Used by media gateway controllers (MGC, also known as call agents) to control media gateways (MG).
• MGCP is based on a master/slave paradigm in which MGC is the master that issues commands to the MG (slave).
IRIS
ET
MGCP- Media Gateway Control ProtocolMGCP has its two earlier protocols: • Simple Gateway Control Protocol (SGCP) and Internet Protocol Device
Control (IPDC).• MGCP uses Session Description Protocol (SDP) to describe the media
sessions. SDP describes session parameters of the media flow between theMGs such as IP addresses, the User Datagram Protocol (UDP) port, RTPprofiles, and multimedia conference capabilities.
• MGCP follows the conventions of SDP as defined in RFC 2327, andimplementations are expected to conform. The SDP specification definesseveral media types:
• IP Addresses—Specify remote gateway, local gateway, or multicast audioconference addresses used to exchange RTP packets
• UDP Port—indicates the transport port used to receive RTP packets fromthe remote gateway
• Audio Media—Specify audio media, including codec
IRIS
ET
MGCP MODELEndpoints:
• Endpoints are sources or sinks of data. They are either physical or logical entities thatexist in an MG. Trunk circuits connecting gateways and telephone switches are physicalendpoints.
Connections:
• Connections can be either point-to-point or multipoint form. A Point-to-point connectionis an association between two endpoints with the purpose of transmitting data betweenthem. Multipoint connections connect an endpoint to a multipoint session.
Calls
• A group of connections composes a call. Call agents assign call identifiers, which areunique for each call and are globally unique throughout the system.
IRIS
ET
MGCP Commands and Messages
Basic call Control Commands
• CreateConnection (CRCX) - Call agents use this command to create connection between end point within a gateway.
• ModifyConnection (MDCX) – The ModifyConnection function changes the characteristics of the gateway's view of a connection or call.
• DeleteConnection (DLCX) – A call agent or a gateway issues the DeleteConnection function to terminate a connection. IR
ISET
MGCP Commands and Messages
Advanced call Control Commands
• NotificationRequest (RQNT) - The NotificationRequest command advises the gateway to notify the originator when a specified event occurs in an endpoint.
• Notification (NTFY) - The gateway sends a Notification based on requested events in the notification request and on the occurrence of these observed events.
IRIS
ET
MGCP Commands and Messages
Management commands
• AuditConnection (AUCX)- Call agents use the AuditConnection command to retrieve information about connections.
• AuditEndpoint (AUEP)- The call agent can use the AuditEndpointcommand to determine the status of an endpoint.
• RestartIn-Progress (RSIP)- The gateway uses the RestartIn-Progress command to inform the call agent that an endpoint or group of endpoints was taken out of service or is back in service.
• EndpointConfiguration(EPCF)- EndpointConfiguration commands enable the call agent to specify the encoding of signals received by the endpoint.
IRIS
ET
MGCP Commands and Messages
MGCP Response Messages –
All MGCP commands are acknowledged. The acknowledgement carries a return code, which indicates the status of the command. The return code is an integer number for which four ranges of values have been defined:
• Values between 100 and 199 indicate a provisional response.
• Values between 200 and 299 indicate a successful completion.
• Values between 400 and 499 indicate a transient error.
• Values between 500 and 599 indicate a permanent error. IR
ISET
MGCP Call Flow
IRIS
ET
SCCP- Skinny Client Control Protocol
• The Skinny Client Control Protocol (SCCP), or Skinny, is a Ciscoproprietary signaling protocol for VoIP devices.
• it is used for registration of endpoints; call messages such as off-hook,on-hook, addressing (dial numbers); and controlling phone display.
• it is a lightweight protocol and, if you have the right dissector, isstraightforward to read and understand.
• When combined with the light weight Cisco call manger called CallManager Express (CME), small VoIP deployments supporting anumber of features and phones can be set up quickly.
• It is designed to be used with a call server and communicates ontransmission control protocol (TCP) port 2000
IRIS
ET
SCCP- Skinny Client Control Protocol
• The Skinny Client Control Protocol (SCCP), or Skinny, is a Ciscoproprietary signaling protocol for VoIP devices.
• it is used for registration of endpoints; call messages such as off-hook,on-hook, addressing (dial numbers); and controlling phone display.
• it is a lightweight protocol and, if you have the right dissector, isstraightforward to read and understand.
• When combined with the light weight Cisco call manger called CallManager Express (CME), small VoIP deployments supporting anumber of features and phones can be set up quickly.
• It is designed to be used with a call server and communicates ontransmission control protocol (TCP) port 2000
IRIS
ET
IAX (The “Inter Asterisk eXchange” Protocol)
• Asterisk comes when you have to pronounce the name of thisprotocol.
• Newbies say “eye ay ex”; those in the know say “eeks.” IAX* is anopen protocol, meaning that anyone can download and develop for it.
• IAX is a transport protocol (much like SIP) that uses a single UDP port(4569) for both the channel signaling and Real-time TransportProtocol (RTP) streams.
• IAX is a binary-encoded protocol.IRIS
ET
UNISTIM (Unified Networks IP Stimulus)
• Telecommunications protocol developed by Nortel (now acquired byAvaya) for IP Phone (terminals and soft phones) and IP PBXcommunication
• The protocols works through a "master" / "slave" mode of operations.
IRIS
ET
MEDIA TRANSPORT PROTOCOLSRTP (Real-time Transport Protocol)
• RTP supports the transfer of real-time media (audio and video) overpacket switched networks.
• It is used by both SIP and H.323.• The transport protocol must allow the receiver to detect any losses in
packets and also provide timing information so that the receiver cancorrectly compensate for delay jitter.
• The functions provided by RTP include:• Sequencing• Payload Identification• Frame Indication• Source Identification• Intermedia Synchronization
IRIS
ET
RTP (Real-time Transport Protocol)
IRIS
ET
RTCP (Real-time Control Protocol)
• RTCP is a control protocol and works in conjunction with RTP.
• In a RTP session, participants periodically send RTCP packets to obtainuseful information about QoS etc.
• The additional services that RTCP provides to the participants are
• QoS feedback
• Session Control
• Identification
• Intermedia SynchronizationIRIS
ET
C-RTP(compressed Real-time Transport Protocol)
• To reduce the large percentage of bandwidth consumed by a G.729 voicecall, you can use cRTP.
• cRTP enables you to compress the 40-byte IP/RTP/UDP header to 2 to 4bytes most of the time.
• cRTP uses some of the same techniques as Transmission Control Protocol(TCP) header compression.
• In TCP header compression, the first factor-of-two reduction in data rateoccurs because half of the bytes in the IP and TCP headers remain constantover the life of the connection.
• Therefore, the algorithm can simply add 1 to every value received. Bymaintaining both the uncompressed header and the first-order differencesin the session state shared between the compressor and the decompressor,cRTP must communicate only an indication that the second-orderdifference is zero.
IRIS
ET
C-RTP(compressed Real-time Transport Protocol)
• In that case, the decompressor can reconstruct the original headerwithout any loss of information, simply by adding the first-orderdifferences to the saved, uncompressed header as each compressedpacket is received.
IRIS
ET
RUDP: Reliable User Data Protocol (RUDP)
• It builds in some reliability to the connectionless UDP protocol.
• RUDP enables reliability without the need for a connection-basedprotocol such as TCP.
• The basic method of RUDP is to send multiples of the same packetsand enable the receiving station to discard the unnecessary orredundant packets.
• This mechanism makes it more probable that one of the packets willmake the journey from sender to receiver.IR
ISET
Stream Control Transmission Protocol
• SCTP is a relatively new transport protocol.
• It is a standard of IETF - RFC 2960
• It was designed to address the shortcomings of both TCP and UDP,especially as related to the types of services that used to be deliveredover circuit switched telephony networks.
• Some of the goals of SCTP were:
• Better congestion avoidance techniques (specifically, avoiding Denialof Service attacks)
• Strict sequencing of data delivery
• Lower latency for improved real time transmissions
IRIS
ET
Stream Control Transmission Protocol
• Reservation protocol, MPLS contains no method to dynamicallyestablish LSPs, but you can use the Reservation protocol (RSVP) withMPLS.
• RSVP is a signaling protocol used to simplify the establishment of LSPsand to report problems to the MPLS ingress router.
• The advantage of using RSVP in conjunction with MPLS is thereduction in administrative overhead.
• If you don’t use RSVP with MPLS, you’ll have to go to every singlerouter and configure the labels and each path manually.IR
ISET
Secure Real-time Transport Protocol (SRTP)
• Secure Real-time Transport Protocol (SRTP) is a profile of the real-time Transport Protocol (RTP).
• SRTP provides integrity, authenticity and privacy protection to the RTPtraffic and to the control traffic for RTP, RTCP.
IRIS
ET
CHAPTER - 3 ASTERISK BASED PBX
IRIS
ET
Open Source IP Telephony Software • Open source software provides users the freedom of choice.
• It eliminates vendor lock-in as well as promotes openness andstandardization.
• There are various successful open-source IP Telephony software thatcan be utilized for the overall benefit of Indian railways voicemodernization.
Asterisk: It is one of the oldest open source telephony software and isvery mature.
• It enjoys a lot of commercial support from the industry. The softwareSupports SIP and is quite flexible.
• It also have support for functioning as a gateways and doestranscoding. It supports variety of encoders including G.711, GSM andG.729.
IRIS
ET
Open Source IP Telephony Software OpenSIPS:
• It is a robust and performant SIP (RFC 3621) Registrar server, Locationserver, Proxy server and redirect server.
• It is modular and has a small foot-print and hence is not resourcehungry.
• It can do both stateless and transactional state full SIP proxyprocessing. It supports IPv4 as well as IPv6, SRV and NAPTR DNS, SRVDNS failover, ENUM, database backend etc.
FreeSwitch:
• FreeSwitch is a scalable open source cross-platform telephonyplatform designed to route and interconnects popular communicationprotocols using audio, video, text or any other form of media.
IRIS
ET
Open Source IP Telephony Software Yate: Yate stands for Yet Another Telephony Engine.
• Its developers call it the next-generation telephony engine.
• Its power lies in its ability to be easily extended. Voice, Video, Data andinstant messaging can all be unified under Yate’s Flexible routingengine, maximizing communications efficiency and minimizinginfrastructure costs.
IRIS
ET
Advantages • No license fees: The Open Source model ensures that open software
remains free. Meaning, an Open Source telephony software solution (e.g.Asterisk telephone systems) in itself and therefore costs you nothing.
• Cross-system usability: Open source invariably ensures Open Standards, itis entirely possible that both hardware and software from multiplemanufacturers are then interoperable with the phone system, giving youmore choice.
• Collaboration: software such as Asterisk telephone systems are developedby variety of developers, bringing not only innovation but also the desire todeveloper not only the highest quality but also generally useful solution.
• Quick reaction times: As so many people globally are involved in thedevelopment, errors are often quickly uncovered and corrected.
• Independence: The server and telephony hardware for an Asterisktelephone system and other open systems can be sourced from a numberof manufacturers, who also utilize open standards. As such, Open Standardphone system users have more freedom when choosing their telephones,server and gateways.
IRIS
ET
Disadvantages
• Support options: Not every VoIP phone system offers you the benefitof professional support. It could be the case that you might have tosolve problems on your own or rely of voluntary help.
• Uncertain development: Open Source project developers are not ascoherently organized as their counter parts in the proprietary world.Depending on the system, reliable new feature innovationimplementation within the phone system can take longer.
• Irregular Updates: A freely or ad hoc developed VoIP phone systemmay not benefit from as frequent updates as you may like.IR
ISET
Asterisk
• Asterisk is an open source software implementation of a PrivateBranch Exchange (PBX).
• The software was originally created in 1999 by Mark Spence ofDigium Corporation in the United States.
• Digium is primary developer for the Asterisk software package as wellas a range of associated software and hardware products.
• Asterisk was designed for the Linux operating system and can beinstalled on either PC servers or compatible embedded hardware.Asterisk is available in a range of different formats and licenses.IR
ISET
Features and Performance of Asterisk PBXFeatures
• CTI
• Audio codecs ADPCM, G.711, G.722, G.72 & GSM
• VoIP Protocols SIP, IAX & MGCP
• Traditional Telephony Protocols E&M, FXS, FXO, Loop start, MF andDTMF
• ISDN Protocols AT&T, Euro ISDN BRI/PRI, Lucent SESS, Nortel DMS100,QSig IR
ISET
Features and Performance of Asterisk PBXPerformance of an Asterisk System
• Number of Concurrent Calls.
• Number of Concurrent Calls.
• Use of complex codec's such as G729, G723.1 and GSM.
• Echo Cancellation.
• Hardware for Analog connections.
• Reliability and Scalability requirement.
• Underlying LAN/IP Network.IRIS
ET
Server Specifications for 100 Lines• Server should be suitable for 24x7 operations.
• Server should be from reputed brand like Dell, IBM and HP.
• Server should be installed in 1+1 redundancy.
• Processor should be Intel Atom dual core with minimum 1.8 GHz, 2GB RAM, 256 MB cache with RAID1 dual HDD 250 GB minimum.
• Operating system should be recent standard Linux distribution.Server should work on 48V DC or 230V AC supply.
• Server should have dual Ethernet interfaces.IRIS
ET
Server Specifications for systems with up to 1000 to 1200 subscribers for
handling 500 to 600 concurrent calls
• Server should be suitable for 24x7 operations.
• Server should be from reputed brand like Dell, IBM and HP.
• Processor should be quad core Xeon processor with minimum 8 GBRAM, 512 MB Cache, RAIDs HDD 500 GB Minimum.
• Server should be installed in 1+1 redundancy. The second server maybe provided at geographically different location.
• Operating system should be recent standard Linux distribution.Server should work on 48V DC or 230V AC supply.
• Server should have dual Ethernet interfaces.
IRIS
ET
Analog Gateway Specifications • It should be an IP to TDM gateway.
• It should work both in registration and point to point mode.
• It should support SIP protocol with RFC 3261, RFC 3262, RFC 3263 &RFC 3264.
• It should support encryption of signaling like TLS and SRTP.
• It should support protocols like G.711, GSM, G.729, iLBC, G.722codec’s
• It should support protocols like T.38, T.30 for fax operations.
• It should have dual Ethernet interfaces.
• It should have GUI based configuration option.
IRIS
ET
Trunk Gateway Specifications • It should be with four RJ45 slots on a single chassis.
• It should be managed by telnet, serial or web.
• It should have dual Ethernet interfaces.
• It should support protocols like SIP, SS7 and QSig.
• It should support protocols like G.711, GSM, G.729a/b, iLBC, G.722codecs.
• It should support protocols like T.38, T.30 for fax operations.IRIS
ET
Other Specifications SIP phones Specifications
• SIP phones should be as per TEC GR No.TEC/GR/SW/TER/SIP/02/Mar'2010.
Telephone connectivity
• Subscriber having LAN accesses are provided with IP telephones.
• Analog telephones/fax is provided through analog gateways.
• Remote locations are connected on WAN or E1 link.
• Remote locations not having IP or LAN connectivity are connected withFXO/FXS pairs on PDH/SDH Mux.
Inter exchange connectivity
• TDM exchanges are connected on E1/PRI interface with suitable gateways.
• IP exchanges are connected on IP trunk interface over a WAN or Ethernetlink.
IRIS
ET
Network requirements To ensure best possible voice quality and reliability, following isrecommended.
• Guaranteed bandwidth to be provided for voice traffic or priority tovoice traffic should be established using QoS tools. This QoS tool canalso be integrated in MPLS network by assigning MPLS labels.
• VLAN's should be used to separate unnecessary broadcasts in thevoice network.
• LAN should have sufficient redundancy so that failure of any singleswitch does not isolate very important telephones or the sever. Veryimportant telephones and the Asterisk server should haveredundancy at the access layer also i.e., they should have connectionto two different switches.
• Switches should have PoE to avoid separate power supply for each IPtelephone.
IRIS
ET
Security Considerations • TLS and SRTP should be used for signaling and media security.
• Voice network should be provided with firewalls and VLANs. A mechanism to allow VoIP traffic through firewalls is required.
• Only authentic devices should be allowed to access the network. Unnecessary software should be removed from server.
• Unnecessary services should be disabled.
• Keep operating system up to date. User Accounts should have very strong passwords.
• Only the required network ports should be opened. Regular backups should be made.
• Ensure physical server security
IRIS
ET
Chapter 4NGN
NEXT GENERATION NETWORK
IRIS
ET
NGN
• ITU-T Y.2001 defines NGN as follows.
"A Next Generation Network (NGN) is a packet-basednetwork able to provide services includingTelecommunication Services and able to make use ofmultiple broadband, QoS-enabled transporttechnologies and in which service-related functionsare independent from underlying transport-relatedtechnologies. It offers unrestricted access by users todifferent service providers. It supports generalizedmobility which will allow consistent and ubiquitousprovision of services to users."
IRIS
ET
IRIS
ET
IRIS
ET
NGNToday Tomorrow
Telephone
network
Mobile radio
network
IP-Network
Multimedia Access - Advantages:
• easy to handle
• reliable
• mobile
Internet
IRIS
ET
IRIS
ET
IRIS
ET
ADVANTAGES
• One network supports transmission of voice, data and video.
• No need to maintain two types’ networks viz., one for voice and one for data.
• Maintenance costs are less as no need to go for a variety of devices.
• Configuration process is simple can be done from a central location Quick deployment of new services.
• Supports all existing PSTN services like call forwarding and do not disturb.
• Number portability; phone number can be retained while changing ISP.
• Supports IP TV a growing business trend.
• One backbone for voice and data services instead of two parallel ones.
• No maintenance of proprietary switching systems.
• Fewer call controlling entities in the network so less capital and operating cost.
IRIS
ET
MumbaiKolkatta
Chennai
Delhi
Ambala
KanpurPatna
Ahmedabad
Bhopal
Secunderabad
Visakhapatanam
Bangalore
Pune
Intermidiate
Router
Intermidiate
Router
Madurai
Trivandrum
Ernakulam
Goa
Surat
Vadodara
Jaipur
Intermidiate
Router
NagpurBilaspur
RaipurTatanagar
Bhubneswar
Lucknow
JalandharRoorkee
Chandigarh
Intermidiate
Router
AllahabadIndore
Kharagpur
Jabalpur
Varanasi
RailTel IP-MPLSBackbone Network
Intermidiate
Router
Intermidiate
Router
Coimbatore
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiat
e Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Intermidiate
Router
Core- 1 Router-M-20
Router-M-10
Core -II Router-M-20
Intermidiate
Router
Intermidiat
e Router
Intermidiate
Router
STM1 Fast Ethernet
E1
E3
Ludhiana
GE
Intermidiate
Router
Guwahati
Intermidiate
Router
BZATPTY
Router-J2350
IRIS
ET
Prepared By-Ajay Pratap Singh
IRIS
ET
ARCHITECTURE
• The architecture basically comprises of :• Network Elements needed for the provision of traditional Telephony services.
• NGN has a layered architecture.
• The layers of NGN :• Access layer,
• Core layer or Transport Layer,
• Control layer and
• Service & Application layer.IRIS
ET
ARCHITECTURE
IRIS
ET
ARCHITECTURE
•Each element :•has distinct roles within the network,•designed to integrate :•horizontally with other elements in the same
layer,•vertically with the function-based elements of
the other layers.IRIS
ET
Packet Network
WAN UTRAN
Mobile PSTN
COCableDSL
Broadband
WLLAccess
Edge
Core
Control
Applications
Management
Content Media
Gateway
Management system
ResidentialUsers
Remote Office/SOHOEnterprise Customers MobileUsers
Softswitches
Application Servers
IRIS
ET
AG MG SG
SIP-T / BICC
DSS1 over SCTP /
SIGTRANSS7 over
SCTP / SIGTRAN
MEGACO/
H.248
Media
Server
Appl.
Server
H.323
/ SIP
SIP
ProxySIP
SIP
SIP
SIP /
SIP-T
SIP
MG SG
PLMNPSTN
PABX
SS7SS7TDMTDMDSS1
GSMSoft
Phone
ENUM
Server
MGC MGC
SIP
Telephone
BB-
RAS
Fire-
wall
SIP
BIND
Packet
Network
H.323
H.323-
Network
AGAG MGMG SGSG
SIP-T / BICC
DSS1 over SCTP /
SIGTRANSS7 over
SCTP / SIGTRAN
MEGACO/
H.248
Media
Server
Media
Server
Appl.
Server
Appl.
Server
H.323
/ SIP
H.323
/ SIP
SIP
Proxy
SIP
ProxySIP
SIP
SIP
SIP /
SIP-T
SIP
MGMG SGSG
PLMNPSTN
PABX
SS7SS7TDMTDMDSS1
GSMSoft
Phone
ENUM
Server
MGC MGC
SIP
Telephone
BB-
RAS
Fire-
wall
SIP
BIND
Packet
Network
H.323
H.323-
Network
H.323-
Network
IRIS
ET
NGN layers
•The Next Generation Networks architecture is based on four layers:➢Access layer,➢Core or transport layer,➢Control layer and➢Application & Service layer.➢Management layer
IRIS
ET
NGN layers
1. Access layer includes :• Traditional networks;
• PSTN, ISDN, PLMN , CABLE …• Specialized packet networks.
❖ Access layer elements include different Media Gateways that support connection to and from the access network with the core network. IR
ISET
NGN layers
2. Core Or Transport layer is the network handling converged services based on IP. • Includes high capacity switches and routers, in
addition high capacity links.• Packet network , VoIP, Media gateways and
Signaling gateways
3. Control layer is the Call Server that provides:• Call Control functions (Soft switch , Call Agent,
Gatekeeper• The Control of the Media Gateway. ( MGC)
IRIS
ET
NGN Control Layer
• Provide capabilities of:• Call Control, • Connection Control,• Protocol Handling and• Other management issues
• The primary part of this layer is: Softswitch.IRIS
ET
NGN Ctrl Layer
❖ Softswitch• Is the core of NGN,
• Independent of Transport Layer.
• Main functions are included:– Call control– Resource distribution– Protocol handling– Routing
– Authentication and – Charging
IRIS
ET
NGN Ctrl Layer
•handles signaling and management entities.
•Never concerns with detailed routingprocedures and controlling network partial components.
• its other tasks are:•Providing Security for connections and•Network Management.IR
ISET
NGN layers
4. Application & Service layer :• plays the role of an IN-SCE (Intelligent Network
Service Creation Environment) extending their functionality in order to cover the new network scenarios.
• Application server• Features server• Media server
IRIS
ET
NGN layers
5. Management layer
•Resource management (capacity, ports, and physicalelements) and QoS in access to the network and thetransport network.
•Various media processing, encoding, data transfer(information flows)
•Management of calls and connection. Managementand interworking of all elements of the referencearchitecture
•Service controlIR
ISET
NGN Components
•The Softswitch
•The Packet core Network
•The Access Networks
•The Media & signaling Gateways
•The Call Server
•The application Server
•The Application Creation Environment
IRIS
ET
Media Server in a SoftswitchArchitecture
Media Server
MGCP,Megaco,
SIP,VoiceXML
RTP
RTP
VoATM
MGCP/
Megaco
Traditional
Phone
SoftswitchApplication
Server
Media Gateway
(xDSL, cable, PSTN,
wireless)
MGCP/Megaco
H.323/SIP
H.323/SIP/MGCP
Basic Services Enhanced Services
IP Phone
IRIS
ET
Signaling
GatewayIP/ATM
SS7
Network
SS7 over IP
SIGTRAN / TALI / Q.2111
MTP2
MTP3
Protocol
MTP2
MTP3
IP
UDP
IP
UDP
Protocols
SS7
DeviceSS7 - > IP protocol Translation
Protocols
IP Telephony
Application
Protocol
SIGNALLING GATEWAY
IRIS
ET
SIGNALLING GATEWAY
• Signalling network: provide change of signalization systems between PSTN/PLMN to VoIP
IRIS
ET
MEDIA GATEWAY
❑ Provides Translations between circuit switched networks and packet switched networks.
❑ Sends notification to the call agent about endpoint events.
❑ Execute commands from the call agents.IRIS
ET
Media Gateway
❑Media Gateway Functionality
•Bearer Interworking Function• Interworking Between Multiple Interface Protocols:
ATM,TDM, Frame Based (IP, FR)•QoS, Traffic and Congestion Management
•Congestion Management•Using Priorities based on Traffic ParametersIR
ISET
Media Gateway
•Different QoS For Different Services
•Traffic Policing per Connection
•Traffic Shaping per Connection•Flat Shaping•Hierarchical Shaping (shaped VCs in shaped VPs) IR
ISET
Media Gateway Architecture
IRIS
ET
MG Types
According to capacity and access level to the core network, Media Gateway is categorized to 3 groups:
• Trunking Gateway: interface between the PSTN/PLMN and VoIP
• Access Gateway :provide traditional analog or PBX interface to VoIP
• Residential Gateway :provide traditional analog(RJ11) interface to VoIPIR
ISET
MEDIA GATEWAY CONTROLLER
❖ Media Gateway Controller Functionality
• Provides End-to-End Call Control
• Supports Call Control Signaling (ISUP, BICC, IN/TCAP, ISDN)
• Supports Signaling Interworking Between Different Signaling Protocols (e.g., ISDN-ISUP-BICC)
IRIS
ET
MEDIA GATEWAY CONTROLLER
•Correlates Between Call Control Signaling and Bearer Control Signaling (BICC)
•Communicates With Feature Servers to Determine Service and Some Call Parameters (TCAP)
•Coordinates Call Progress and Resources Management with the Bearer Control Function (H.248/MEGACO)
IRIS
ET
Media
Gateway
Controller
SIP-BCP-T
H.323
MEGACOMGCP
SIGTRAN
TALI
SIP
Control SwitchCall Agent Media Gateway Controller (MGC)
Call
Handling
MTP2IP
UDP
SIGTRAN
PSTN Switch Media Gateway Controller
Signaling Gateway
ProtocolMTP3
MTP2
MTP3
ISUP
IP
UDP
SIGTRAN
Protocol
ISUP
Call
HandlingProtocol Translation
and addressing
Media Gateway Controller
IRIS
ET
Servers
•Application Server
•Media Server
•Call Server
•Feature Server IRIS
ET
Application Server
App. Server functionality :
•Application server is implemented to performfunctionalities specific to certain service, performspecialized service logic call control, also includesmore functionalities in terms of user web interface,endpoints management, etc. For example it canprovide specific videoconferencing service, CallCenter service or IP Centrex service
•Billing services
•VPN
•Calling Card services
• IN services
IRIS
ET
Media ServerMedia Server functionality :
• Media server provides functionalities allow interaction between callingparty and application using end-point device. It provides Media ResourceFunctions (tones detection, speech synthesis and recognition,compressions, media mixing, etc.) and Media Control Functions that controlof media functions (voice message play management, conference bridge,fax message management, etc.)
•Voice Mail services•Fax Mail Box•Voice Recognition•Video Conferencing•Voice to Text•Unified Messaging•Fax over IP by means of T.38
IRIS
ET
IRIS
ET
End Point Connection
IRIS
ET
Related Documents
![[VoIP] Cisco CallManager Basics (VoIP)](https://static.cupdf.com/doc/110x72/55cf99c3550346d0339f0d92/voip-cisco-callmanager-basics-voip.jpg)
