Communication Systems - 4 th Lecture Large scale telephony networks IN and Asterisk

Transcription

1 - 4 th Lecture Large scale telephony networks IN and Asterisk Chair in Communication Systems Department of Applied Sciences University of Freiburg 2010

2 Course Information Slides and Recordings You will find the course material online: Lecture recordings and PDF of the slides some when after the lecture on the electures portal of the faculty: Slides and exercise sheets at the lectures home page: for each topic Slides are produced in a just-in-time fashion, but we try to upload the final version Thursday in the evening or latest just before the lecture 11/12/10 2

3 Last lecture introduction to telephony networks Last lecture ISDN Integrated Services Digital Network First fully digital telephony network Signaling between terminal endpoints (TE) and the local switching center matched to OSI layers BRI offers two B channels (64kbit/s for either voice or data communication) and a separate D channel for out of band dial and control signaling D channel are defined in the 3 lower OSI layers physical interface with different encodings, e.g. 4B3T DLL represented by LADP protocol 11/12/10 3

4 Last lecture introduction to telephony networks Last lecture DSS1 is handling call setup, signaling, call destruction, gives information on caller ID, Seen call setup, signaling, PCM channels in several packet dump and analyzing exercises Recapitulate ISDN is a circuit switched network Circuit path between two parties reserved for the duration of the call Released upon disconnect Function Groups & Reference points (LT,TA,NT,S,U,TE1,TE2) 11/12/10 4

5 Plan for this lecture Today we will talk about Intelligent Networks as a higher level feature implementation Signaling in large scale digital networks Primary Rate Interface as example for modulation But how is a call setup and routed globally Q.SIG for inter-connecting PBX Introduction to Asterisk as a software PBX implementing most of the features and protocols we are talking about in this lecture 11/12/10 5

6 Primary Rate Interface - PRI Talked on ISDN Basic Rate Interface (BRI) last lecture Enough for average household or small office But insufficient to serve larger enterprises and organizations Primary Rate Interface (PRI) handles large scale connectivity ITU-T specifications G.703, G.704, G.705 includes 30 B channels (each B channel 64kbit/s), a full rate D channel at 64kbit/s and a framing/synchronization pattern (64kbit/s) Channels could be bundled, so called H channels: H0 384kbit/s, H kbit/s and H kbit/s 11/12/10 6

7 PRI physical specification, interfaces Brutto bandwidth sums up to 2.048kbit/s connection Copper wire connections allow up to 250m without refresh, for longer distances often fiber optics All connections are unidirectional, so no channel separation is needed as was in BRI Name of the interface from switching center: U K2, for fiber optics: U G2, user interface is named S 2M Different channels transmitted in TDM (Time Division Multiplexing) International E1 (European ISDN standard) systems use HDB3 (High Density Bipolar 3) line coding 11/12/10 7

8 PRI HDB3 line coding Two kinds of transmission media used to transmit E1 signals Coaxial cable (2,37V peak base) Twisted-pair cable (3V peak base) HDB3 Line Coding is similar to AMI coding To avoid co-current flow strings of 4 zeroes are replaced with one of four bipolar violation codes, example for 8 consecutive zeros 11/12/10 8

9 Network signaling Options to bundle more than 30 channels into a site connection available other media, e.g. fibre optics is used then (underlying transport technology is often ATM Asynchronous Transfer Mode) In BRI and PRI connections a separate channel is used for signaling D / DSS1 multiplexed into the same physical connection In modern large scale telephony networks signaling and real connections are completely independent Connectivity between switching centers is handled by a specialized signaling system Signal System Nr. 7 (SS7, next lecture) 11/12/10 9

10 Independent networks for signaling and connection Signaling layer consists of Signaling Points (SP) and Signaling Transfer Points (STP) 11/12/10 10

11 Line switching and signaling network Signaling layer is a virtual layer on-top the connection layer network (coupling network) SP's are the direct involved switching centers of a connection, STP just route signaling information End points of a connection are the the end switching centers where the subscribers are connected to Every switching center should be connected at least to two STP for backup 11/12/10 11

12 Line switching and signaling network SP's are the direct involved switching centers of a connection, STP just route signaling information Thus route optimizations and fall-back routes implemented Signaling data itself is transported in usual bearer channels (B channel) of the connection layer of the switching centers 11/12/10 12

13 Line switching and signaling network Signaling network uses a network indicator to distinguish different networks SP use a Signaling Point Code of 14bit and could be associated with up to four networks This allows changeovers between different networks For international connections special ISTP (International Signal Transfer Points) are operated 11/12/10 13

14 Signaling between Private Branch Exchanges - Q.SIG By now talked of intra- and inter provider signaling networks But what to do for large organization, company telephony systems like this university Typically internally handled for a number of reasons: Costs, privacy/security, efficiency (number of internal vs. number of external calls) Proprietary features of every equipment provider typically prohibits to couple different PBXes In-acceptable for large organizations: Company mergers or split would require a completely new telephony system etc. Special inter PBX protocol defined: Q.SIG 11/12/10 14

15 Signaling between Private Branch Exchanges - Q.SIG Digital PBX are connected to the public network most commonly via ISDN BRI or PRI interfaces Internally they use not SS7 but DSS1 (D channel protocol) in small scale exchanges and QSIG in large scale PBX Q.931 protocol was intended to be used for signaling within PBX but every manufacturer created his own protocol (there is much money in the market and thus much interest that a customer does not uses different equipment) But at least a common subset of features exists... 11/12/10 15

16 Signaling between Private Branch Exchanges - Q.SIG QSIG was specified by the ECMA (European Computer Manufacturers Association) Q in the name refers to the Q reference point in the PBXs At layer 1&2 QSIG is identical to the DSS1 EURO ISDN protocol The layer 3 is split into three sub-layers: Basic Call (BC), Generic Function (GF) Protocol and QSIG Procedures for Supplementary Services (SS) BC implements ISDN standard functionality, GF should allow the inter-connect of devices of different vendors, SS allows for transparent services extensions (automatic call completion, display of tolls,...) 11/12/10 16

17 Signaling between Private Branch Exchanges - Q.SIG Based on Q.931, Q.SIG is an evolving technology that allows communication with legacy PBX and key systems Q.SIG could be run over IP networks (thus easily linking pretty remote sites without the need for a separate typically expensive infrastructure) 11/12/10 17

18 Higher layer functionality Switching, call routing and a number of basic ISDN features are not enough for modern telephony networks, especially with the implementation of mobile telephony Intelligent Network (acronym IN) network architecture for fixed as well as mobile telecommunication networks Abstract higher layer (intra/inter provider) service Major driver behind the development of the IN system need for a more flexible way of adding sophisticated services 11/12/10 18

19 Intelligent Networks Provides services in addition to the standard telecommunication services such as PSTN, ISDN, GSM, UMTS,... services Intelligence provided by network nodes on the service layer Distinct to the switching layer of the core network Operated at higher layers above core switches or telephone equipment IN nodes typically operated/owned by telecommunications operators 11/12/10 19

20 Intelligent Networks Examples of IN services (stuff telephony operators sell to you, especially for mobile phone contracts and prepaid services) Call screening Telephone number portability Toll free calls / Freephone Prepaid calling Account card calling Private-number plans (with numbers remaining unpublished in directories e.g. for politicians, psychiatrists,...) 11/12/10 20

21 Intelligent Networks Services Universal Personal Telecommunication service 0700 prefix in many countries (e.g. Germany) Universal personal telephone number independent of the local prefix, transferable Anybody here owning/using such a telephone number!? Mass-calling service Prefix free dialing from cellphones abroad Reverse charging Home Area Discount (offered by many mobile phone operators) 11/12/10 21

22 Intelligent Networks Services Call distribution based on various criteria associated with the call Location Based Routing Time based routing Proportional call distribution (e.g. between two or more call centres or offices) 11/12/10 22

23 Intelligent Networks Services IN based on the Signaling System #7 (SS7) protocol (introduced next lecture) Running between telephone network switching centers and other network nodes owned by network operators IN concepts, architecture and protocols originally developed as standards by the ITU-T Defined by standards Q.1210 to Q.1219, aka Capability Set One (CS-1) Complete architecture including the architectural view, state machines, physical implementation and protocols 11/12/10 23

24 PBX implementation Asterisk Telephony equipment incredible expensive (just ask for the purchase price and operating costs of the Siemens ISDN PBX used in the university) Even small scale equipment not easily affordable for end users We do not have such equipment for experiments either, but there are good alternatives Asterisk is an Open Source PBX implementation and media gateway Implements most of the stuff we are talking about in this lecture 11/12/10 24

25 PBX implementation Asterisk Asterisk implements switching elements, talked of in the beginning of this lecture and could be seen as an IN implementation offering most of the features mentioned before Bridge between telecommunication history and future Implements SS7, could connect to and drive ISDN channels Can handle SIP, H323,... 11/12/10 25

26 Asterisk an open source telephone system Asterisk is a software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium Inc. A telephone system is responsible for Controlling the internal telephone network Establish connections between the participants Act as a intermediary between internal and external telephone network Provides basic and additional functionalities as listed in the IN part

27 Asterisk an open source telephone system A telephone system Can accept and route incoming calls Act as a voice mailbox Provide automated computer voice-based functions Agent functionality, voice menu systems (run your own banking or whatever system that way) Most of the additional functionalities of traditional telephony switches depend on the hardware, which can become very expensive and user depends on a single manufactures (ask for a SIEMENS voice menu system :))

28 Asterisk an open source telephone system Asterisk implements all this but could be easily extended, programmed to: Different agent functionality, voice menu systems (run your own banking or whatever system that way) Or produce your own Spam Over Internet Telephony (SPIT) Or fight SPIT running touring tests on callers having source suppressed or unknown telephony numbers (practical exercise)

29 Operation principle of Asterisk terminal connection At the terminal connection, Asterisk is able to Control special hardware (ISDN Cards BRI and PRI/E1) as NT (network terminator) or TE (terminal endpoint) Handle special analogous telephone cards Support of SIP or H323 Control the local soundcard or some types of modem ISDN card are supported with understand CAPI 2.0, AVM Fritz, HFC Chip Cologne, Digium,...

30 HFC Chip cards HFC Chip cards (already used in the practical exercises) are very interesting because They can run their own ISDN Bus (provide D- channel protocol) classic mode: BRI Basic Rate IF, 2 B-channels) Are able to connect to "old" ISDN hardware (infrastructure must not be changed) Despite to AVM Fritz / CAPI, the computer can handle more than one HFC Chip card

31 Operation principle of Asterisk telephone switch Asterisk can Handle multiple codes (PCM ulaw/alaw for ISDN, GSM, G.7xx, ) Transfer from codec A to codec B and vice versa Connect two participants, hold calls, establish callbacks Automated or manual redirection of calls Provide conference calls Calling line identification presentation (CLIP)

32 Operation principle of Asterisk telephone switch Asterisk can Ankommende Anrufe auf mehreren Anschlüssen gleichzeitig signalisieren, wer zuerst abhebt hat den Anruf Nach einstellbarer Klingelzeit es auf dem nächsten Anschluss klingeln zu lassen Agentensystem: Anrufe in Warteschlange parken und die Agenten nach bestimmten Mustern zugreifen lassen Beliebige interner Kurzwahlnummern definierbar Beliebige Wahlregeln anhand der gewählten Zielnummer oder der ermittelten Absendekennung anzuwenden Beliebige Zeitprofile

33 Software architecture Asterisk works with modules: every functionality is organized in a module which provides a flexible, extensible architecture Works internal with channels which represent the connections between the different devices Dialplan is the extensible configuration IAX2: lightweight protocol for the communication between Asterisk-Server and Clients

34 Configuration of Asterisk Directory for the configuration files is /etc/asterisk and contains the following files: asterisk.conf: basic parameters like paths to modules, Callfiles, AGI directory (Application Gateway Interface) extensions.conf: contains the dialplan zapata.conf: only important if Asterisk has to connect to HFC ISDN hardware sip.conf: user management for the connected SIP clients iax.conf: configuration data for the connection with different Asterisk Server via IAX channel (practical exercise)

35 Configuration of Asterisk asterisk.conf

36 Sip.conf Contact SIP Provider Manage and maintain uplink SIP connections, e.g. to SIP providers like 1&1, Sipgate, or other Asterisk servers

37 sip.conf mapping uplink SIP channels (cont) Define entries for each SIP user ID Add the capabilities like supported codecs, NAT,...

38 sip.conf local SIP users Add local SIP users and define the capabilities Comsys SIP accounts created this way

39 Configuration of Asterisk sip.conf Configuration options (try it out in the practical part) Type: Context: Every connection has to be defined User: incoming calls Peer: outgoing calls Friend: both directions Context name has to be the same as in the file extensions.conf (where further informations are located)

40 Configuration of Asterisk extensions.conf Central file which contains four parts: 1.context, 2. extensions, 3. priorities, 4. applications Context = Collection of instructions Eg: [incoming]; String 2 {A,, z, -, } Every instruction is active until the next context definition Security through restriction / unblocking Special: [general] and [global]

41 Configuration of Asterisk extensions.conf Extensions = instructions to be performed Syntax: exten => name, priority, application() Name can be number or string Priority step (can exist multiple steps) Application (command to execute) Priority sequentially step Always begins with 1 Increased by 1 Newer Asterisk versions understand N+1

42 Extensions.conf example for priorities

43 Configuration of Asterisk extensions.conf Application Command to perform Answer() Hangup() Playback(<filename>): play gsm/wav file then next step Background(<filename>): play gsm/wav file with "interrupt" function Goto(context, extension, priority): go to Dial(destination, time in secs, optional beh., URL): connect with...

44 Extensions.conf context incoming

45 Extensions.conf block bad numbers Special number corridors like international calls, expensive 018, 019, 0900, are blocked (to avoid ruining the professorship :))

46 Practical Exercise Asterisk Please grab an exercise sheet This exercise mostly deals with Asterisk setup and features Could be done alone or in small groups Please help each other

47 Practical Exercise Asterisk If you are very fast with your exercise sheet Implement a simple touring test for incoming calls If a CallerID (look up the variable on the net) is suppressed/unknown (not in a list or database) ask the caller to do a simple calculation and send the result by DTMF Check the result and put the caller through if correct, block if not This could be easily done as homework too :)