1 / 17

Miroslav Voz ňák VŠB - Technical University of Ostrava Department of Telecommunications

454-319/1: Vo ice over IP. Lecture No. 6 Gateway configuration, FXS, FXO, EM, ISDN PRI and BRI. Miroslav Voz ňák VŠB - Technical University of Ostrava Department of Telecommunications Faculty of Electrical Engineering and Computer Science 17. listopadu 15, 708 33 Ostrava – Poruba

haines
Download Presentation

Miroslav Voz ňák VŠB - Technical University of Ostrava Department of Telecommunications

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. 454-319/1: Voice over IP Lecture No.6 Gateway configuration, FXS, FXO, EM, ISDN PRI and BRI Miroslav Vozňák VŠB - Technical University of Ostrava Department of Telecommunications Faculty of Electrical Engineering and Computer Science 17. listopadu 15, 708 33 Ostrava – Poruba mailto:miroslav.voznak@vsb.cz http://homel.vsb.cz/~voz29 Miroslav Voznak, lecture on H.323

  2. POTS Plain Old Telephone Service – analog tel. network • CO Central Office - public telephone exchange • telco – Telecommunication Company • LEC – Local Exchange Carrier (local teleph. comp.) • IXC – Interexchange Carrier (for long distance) • FXS - Foreign eXchange Subscriber • An FXS interface provides POTS service (BORSCHT) • FXO - Foreign eXchange Office • An FXO receives POTS service, typically from a Central Office • An FXO interface provides on-hook/off-hook indication (loop closure) Miroslav Voznak, lecture on H.323

  3. Subscriber loop • - suitable when conventional phones should be connected directly to VoGWvia an FXS interface • an FXS interface connects directly to a standardtelephone and supplies ring, voltage, and dial tone, but can also be used for PBXinterconnection • a disadvantage is that when calling inward towards the PBX, an extensionnumber can be dialled only as DTMF (Dual-Tone Multi-Frequency) suffix, after a call isestablished and is already accounted for.This type of interface is usually a low-cost solution; • E&M • E&M commonly stands for both Ear and Mouth or recEive and transMit • It allows extension dialling before the conversation begins. • It requires a special interface card forthe PBX, if the PBX is already equipped with this card, this can also be a low-cost solution; Miroslav Voznak, lecture on H.323

  4. ISDN BRI or PRI • ISDN PRI or BRI is the most commonly used interface for interconnection VoIP gateway with PSTN or PBX. • in the case of PBX there is used DSS1 or QSIG, QSIG is a powerful signaling in corporate network • in the case of PSTN interconnecting it’s mostly used DSS1 signaling but for operators peering is required SS7 • Cisco IOS must be higher than 12.1(5) Miroslav Voznak, lecture on H.323

  5. a practical configuration of Cisco Gateway • configure through: • console cable, RS232, terminal: 9600, 8, NO, 1, HW flow control • remote access: telnet (non-secured) or ssh • basic commands: • enable /* switch to enable mode config t /* switch to configuration mode • show run /* show actual configuration wr /* save configuration • CTRL+Z /* cancel shutdown • ? /* list of options , exit /* exit a level , no /* deactivate or remove option • network settings: • # interface FastEthernet0/0 • # ip address 158.196.81.201 255.255.255.0 /* IP address assigning • # ip route 0.0.0.0 0.0.0.0 158.196.81.1 /* default gateway setting Miroslav Voznak, lecture on H.323

  6. How to configure GnuGK with Cisco GW During the registration is sent GRQ a RRQ with name of Gatekeeper as Gatekeeperidentifier (id) and Gateway is registered wit prefixes listed in h323-gateway voiptech-prefixand with its h323-id. # interface FastEthernet0/0 # ip address 158.196.81.200 255.255.255.0 # h323-gateway voip interface # h323-gateway voip id OpenH323GK ipaddr 158.196.81.103 1718 priority 100 # h323-gateway voip h323-id gw1vsb # h323-gateway voip tech-prefix 0 Name of GNU GK submitted in section Main of file /etc/gatekeeper.ini must be thesame as id of Gatekeeper inCiscoGW [Gatekeeper::Main] Name=OpenH323GK Miroslav Voznak, lecture on H.323

  7. check up the state of registration # config terminal # no gateway /* unregister # gateway /* register to GK show gateway # sh gateway /* show status of registration H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1 H.323 service is up Gateway gw1vsb is registered to Gatekeeper OpenH323GK Miroslav Voznak, lecture on H.323

  8. voice settings: # voice rtp send-recv # voice service voip # h323 # h245 caps mode restricted /* prevents DTMF a T.38 indication in H.245 TCS as a optional # h245 tunnel disable /* by default is H.245 tunneling enabled # voice-port 0/0 # compand-type a-law /* default PCM compand type # cptone CZ /* czech control tones # bearer-cap 3100Hz /* default bearer capability Miroslav Voznak, lecture on H.323

  9. ISDN BRI configuration of Cisco GW for Network Side # interface BRI0/0 /*identification of interface in Cisco # no ip address /* because it’s Voice interface # isdn switch-type basic-net3 /* basic-net3 means BRI with DSS1 /* basic-qsig means BRI with QSIG # isdn overlap-receiving /* ability to accept dialing digit by digit # isdn protocol-emulate network /* L3 will operate in network side mode # isdn layer1-emulate network /* L1/L2 will operate in network side mode # isdn incoming-voice modem /* incoming calls are treated as modem calls # isdn send-alerting /* will send ALERTING message # isdn sending-complete /* can include Sending Complete inf. element # isdn static-tei 0 /* TEI=0 is configuration for Point to Point # isdn skipsend-idverify /* do not send TEI verification for User Side # isdn protocol-emulate user /* L3 will operate in user side mode # isdn layer1-emulate user /* L1/L2 will operate in user side mode # as a optinal isdn no skipsend-idverify /* send TEI verification in PMP Miroslav Voznak, lecture on H.323

  10. ISDN PRI configuration of Cisco GW for Network Side #controller E1 1/0 # clock source line primary /* layer L1 clock derived from the line - preferred or # clock source internal /* or clock running internal # pri-group timeslots 1-31 #interface Serial1/0:15 # no ip address /* because it’s Voice interface # isdn switch-type primary-net5 /* primary-net5 means PRI with DSS1 # isdn overlap-receiving /* ability to accept dialing digit by digit # isdn not-end-to-end 64 /* overriding the speed that the network reports # isdn protocol-emulate network /* interface operates in network side mode # isdn incoming-voice modem /* incoming calls are treated as modem calls # isdn send-alerting /* will send ALERTING message # isdn sending-complete /* can include Sending Complete inf. element for User side # isdn protocol-emulate user /* interface operates in user side mode Miroslav Voznak, lecture on H.323

  11. the timers on ISDN interface # isdn overlap-receiving T302 /* T302 timer (in miliseconds) waits before expiring # isdn overlap-receiving terminating-char # /* a terminating character in CPN # isdn t306 /* a timer to disconnect a call after the router sends a disconnect message # isdn timer t309 /* a timer to clear to clear network connection and to release B-channel and Call Reference when data-link disconnection has occured, L2 recovery = SATUS ENQUIRY before T309 expires or RESTART after # isdn t310 /* a timer for Call Proceeding Messages, the router waits before disconnecting a call after receiving Call Proceeding Miroslav Voznak, lecture on H.323

  12. Inbound Dial Peer - configuration Inbound peers are associated to incoming Call legs from Voice network (VoIP) on the voice gateway - POTS /* assume the router in this example receives a call setup with dial-string 59699XXXX from the network where XXXX is an extension and 59699 is a prefix # dial-peer voice 1 pots # destination-pattern 59699.... /* this rule accepts 59699xxxx where xxxx are 4 digits # direct-inward-dial # port 0/0 /* T terminator can be used to suspend this digit-by-digit matching, T refers to the T302 interdigit timer # destination-pattern 59699T Miroslav Voznak, lecture on H.323

  13. Outbound Dial Peer - configuration Outbound dial peers are associated to outgoing Call legs toVoice network (VoIP) on the voice gateway, session target is then used to forward the call /* assume the dial-string is 603121212, after GW receives digit 3, the gateway matches dial peer 20, routes the call to session target and forwards the complete dial-string # dial-peer voice 20voip # destination-pattern 59699.... # codec g711alaw /* use this codec type # session target ras /* if GW is registered with Gatekeeper we can use RAS signaling then or # session target ipv4:195.113.222.2 /* a target is a IP address, i.e. Gateway Miroslav Voznak, lecture on H.323

  14. The Digit Translation Rules The digit translation rules can be used for incoming or outgoing numbers. We achieve a rewriting of CPN (Called number) or CLI (calling number). A Received number 59699xxxx will be translated to 42059699xxxx and begining 0 to 420, any TON (Type of Number) will be translated to UNKNOWN # translation-rule 1 # Rule 0 ^59699.... 42059699 ANY unknown # Rule 1 ^0 420 ANY unknown # dial-peer voice 20 voip /* the rule 1 placed into dial-peer # translate-outgoing called 1 /* rewriting CPN or # translate-outgoing calling 1 /* rewriting CLI # voice-port 1/0:15 /* the rule placed as a global into voice-port # translate calling 2 # translate called 4 Miroslav Voznak, lecture on H.323

  15. The Voice Class Codec We create the class for codec selection preference and assign an identification tag then we apply the voice class codec to a dial-peer. # voice class codec 99 # codec preference 1 g711alaw # codec preference 2 g711ulaw # codec preference 3 g729r8 # dial-peer voice 20 voip /* outbound peer # voice-class codec 99 /* a class codec in inbound peer # dial-peer voice 999 voip /* The voice-class codec can’t be placed into POTS, # voice-class codec 99 /* so the incoming direction is solved a little bit # incoming called-number ..T /* illogicaly Miroslav Voznak, lecture on H.323

  16. Progress Tones Progress tones are transmitted in-band, such as ringback, busy tones and announcements. The indication of in-band tones is controlled by PI (Progress Indicator). PI is an information element included in SETUP, ALERTING, CONNECT, PROGRESS or DISCONNECT. If PI=1 or PI=8 then In-band information is available. When a Setup arrives at the originating gateway with a PI=3, it means that in-band messages are expected. If user doesn’t hear ringback tone we use these solutions: # voice call send-alert /* global configuration # voice dial-peer 1 pots /* but ringback must come ! # progress_ind alert enable 8 # progress_ind progress enable 8 # progress_ind connect enable 8 # progress_ind setup enable 3 # voice dial-peer 20 voip /* it forces to generate in-band ringback # progress_ind setup enable 3 or tone ringback alert-no-pi Miroslav Voznak, lecture on H.323

  17. Thank you for your attentionmiroslav.voznak@vsb.cz Miroslav Voznak, lecture on H.323

More Related