IP Telephony

• Quicknet LineJACK or PhoneJACK analog telephony cards
• ISDN cards
• Voicetronix OpenLine4 (was V4PCI) - 4 analog telephone lines cards
• Zaptel Wildcard X100P IP telephony card - 1 analog telephone line

• Package Management
• ISDN
• AAA

IP telephony, known as Voice over IP (VoIP), is the transmission of telephone calls over a data network like one of the many networks that make up the Internet. There are four ways that you might talk to someone using VoIP:

• Computer-to-computer - This is certainly the easiest way to use VoIP, and you don't have to pay for long-distance calls.
• Computer-to-telephone - This method allows you to call anyone (who has a phone) from your computer. Like computer-to-computer calling, it requires a software client. The software is typically free, but the calls may have a small per-minute charge.
• Telephone-to-computer - Allows a standard telephone user to initiate a call to a computer user.
• Telephone-to-telephone - Through the use of gateways, you can connect directly with any other standard telephone in the world.

Suppoted hardware:

• Quicknet Technologies cards:
  • Internet PhoneJACK (ISA or PCI) for connecting an analog telephone (FXS port)
  • Internet LineJACK (ISA) for connecting an analog telephone line (FXO port) or a telephone (FXS port)
• ISDN client cards (PCI) for connecting an ISDN line. See Device Driver List for the list of supported PCI ISDN cards
• Voicetronix OpenLine4 card for connecting four (4) analog telephone lines (FXO ports)
• Zaptel Wildcard X100P IP telephony card (from Linux Support Services) for connecting one analog telephone line (FXO port)

Supported standards:

• MikroTik RouterOS supports IP Telephony in compliance with the International Telecommunications Union - Telecommunications (ITU-T) specification H.323v4. H.323 is a specification for transmitting multimedia (voice, video, and data) across an IP network. H.323v4 includes: H.245, H.225, Q.931, H.450.1, RTP(real-time protocol)
• The followong audio codecs are supported: G.711 (the 64 kbps Pulse code modulation (PCM) voice coding), G.723.1 (the 6.3 kbps compression technique that can be used for compressing audio signal at very low bit rate), GSM-06.10 (the 13.2 kbps coding), LPC-10 (the 2.5 kbps coding), G.729 and G.729a (the 8 kbps CS-ACELP software coding), G.728 (16 kbps coding technique, supported only on Quicknet LineJACK cards)

The total delay from the moment of recording the voice signal till its playback is the sum of following three delay times:

• delay time at the recording point (approx. 38ms)
• delay time of the IP network (1..5ms and up)
• delay time at the playback point (the jitter delay)

If autodial does not exactly match an item in /ip telephony numbers, there can be two possibilities:

• if autodial is incomplete, rest of the number is asked (local voice port) or incoming call is denied (VoIP)
• if autodial is invalid, line is hung up (PSTN line), busy tone is played (POTS) or incoming call is denied (VoIP)

Some tips for testing balance registers:

• test is sensitive to noise from the phone, so it's recommended to cover mouth piece during it;
• find-best-balance can be interrupted by clear-call command;
• once best balance-registers value is known, it can be set manually to this best value for all voicetronix voice ports, which will use the same telephone line.

There is a possibility to enter some special symbols in lmsn property. Meaning of the special symbols:

• ; - separates pattern entries (more than one pattern can be specified this way)
• ? - matches one character
• * - matches zero or more characters
• [ ] - matches any single character from the set in brackets
• [^ ] - matches any single character not from the set in brackets

This is the so-called "routing table" for voice calls. This table assigns numbers to the voice ports.The main function of the numbers routing table is to determine:

• to which voice port route the call
• what number to send over to the remote party

• If nr=55555, it does not match any of the destination patterns, therefore it is rejected
• If nr=123456, it does not match any of the destination patterns, therefore it is rejected
• If nr=1234, it does not match any of the destination patterns (incomplete for record #0), therefore it is rejected
• If nr=12345, it matches the record #0, therefore number "" is dialed over the voice port XX
• If nr=11111, it matches the record #1, therefore number "1" is dialed over the voice port YY
• If nr=22987, it matches the record #2, therefore number "333987" is dialed over the voice port ZZ
• If nr=22000, it matches the record #2, therefore number "333000" is dialed over the voice port ZZ
• If nr=444, it matches the record #3, therefore number "55444" is dialed over the voice port QQ

• If nr=222 => the best match is the record #4 => nc=44444, vp=KK (note: the 'best match' means that it has the most coinciding digits between the nr and destination pattern).
• If nr=221 => incomplete record #2 => call is rejected
• If nr=321 => the best match is the record #5 => nc=55321, vp=LL
• If nr=421 => matches the record #3 => nc=55421, vp=QQ
• If nr=335 => the best match is the record #5 => nc=55321, vp=LL

• If nr=335 => incomplete record #6 => the call is rejected. The nr=335 fits perfectly both the record #3 and #5. The #5 is chosen as the 'best match' candidate at the moment. Furthermore, there is record #6, which has two matching digits (more than for #3 or #5). Therefore the #6 is chosen as the 'best match'. However, the record #6 requires five digits, but the nr has only three. Two digits are missing, therefore the number is incomplete. Two additional digits would be needed to be entered on the dialpad. If the number is sent over from the network, it is rejected.
• If nr=325 => matches the record #5 => nc=55325, vp=LL
• If nr=33123 => matches the record #6 => nc=33123, vp=MM
• If nr=123 => incomplete record #0 => call is rejected
• If nr=111 => incomplete record #1 => call is rejected
• If nr=112 => matches the record #7 => nc=77112, vp=NN
• If nr=121 => matches the record #3 => nc=55121, vp=QQ

The contents of the CDR (Call Detail Record) are as follows:

• NAS-Identifier - router name (from /system identity print)
• NAS-IP-Address - router's local IP address which the connection was established to (if exist)
• NAS-Port-Type - always Async
• Event-Timestamp - data and time of the event
• Acct-Session-Time - current connection duration (only in INTERIM-UPDATE and STOP records)
• Acct-Output-Packets - sent RTP (Real-Time Transport Protocol) packet count (only in INTERIM-UPDATE and STOP records)
• Acct-Output-Packets - sent RTP (Real-Time Transport Protocol) packet count (only in INTERIM-UPDATE and STOP records)
• Acct-Input-Packets - received RTP (Real-Time Transport Protocol) packet count (only in INTERIM-UPDATE and STOP records)
• Acct-Output-Octets - sent byte count (only in INTERIM-UPDATE and STOP records)
• Acct-Input-Octets - received byte count (only in INTERIM-UPDATE and STOP records)
• Acct-Session-Id - unique session participient ID
• h323-disconnect-cause - session disconnect reason (only in STOP records):
• 0 - Local endpoint application cleared call
• 1 - Local endpoint did not accept call
• 2 - Local endpoint declined to answer call
• 3 - Remote endpoint application cleared call
• 4 - Remote endpoint refused call
• 5 - Remote endpoint did not answer in required time
• 6 - Remote endpoint stopped calling
• 7 - Transport error cleared call
• 8 - Transport connection failed to establish call
• 9 - Gatekeeper has cleared call
• 10 - Call failed as could not find user (in GK)
• 11 - Call failed as could not get enough bandwidth
• 12 - Could not find common capabilities
• 13 - Call was forwarded using FACILITY message
• 14 - Call failed a security check and was ended
• 15 - Local endpoint busy
• 16 - Local endpoint congested
• 17 - Remote endpoint busy
• 18 - Remote endpoint congested
• 19 - Could not reach the remote party
• 20 - The remote party is not running an endpoint
• 21 - The remote party host off line
• 22 - The remote failed temporarily app may retry
• h323-disconnect-time - session disconnect time (only in INTERIM-UPDATE and STOP records)
• h323-connect-time - session establish time (only in INTERIM-UPDATE and STOP records)
• h323-gw-id - name of gateway emitting message (should be equal to NAS-Identifier)
• h323-call-type - call leg type (should be VoIP)
• h323-call-origin - indicates origin of call relatively to the gateway (answer for calls from IP network, originate - to IP network)
• h323-setup-time - call setup time
• h323-conf-id - unique session ID
• h323-remote-address - the remote address of the session
• NAS-Port-Id - voice port ID
• Acct-Status-Type - record type (START when session is established; STOP when session is closed; INTERIM-UPDATE (ALIVE)session is alive). The time between the interim-update messages is defined by the interim-update-interval parameter (if it is set to 0s, there will be no such messages)

• The IP Telephony does not work after upgrading from 2.5.x version - You need to completely reinstall the router using any installation procedure. You may keep the configuration using either the installation program option or the backup file.
• The IP Telephony gateway does not detect the drop of the line when connected to some PBXs - Different regional setting should be used to match the parameters of the PBX. For example, try using uk for Meridian PBX.
• The IP Telephone does not call the gateway, but gives busy signal - Enable the logging of IP telephony events under /system logging facility. Use the monitoring function for voice ports to debug your setup while making calls.
• The IP telephony is working without NAT, but sound goes only in one direction - Disable H323 service port in firewall: /ip firewall service-port set h323 disabled=yes
• The IP Telephony does not work through NAT - Enable H323 service port in firewall: /ip firewall service-port set h323 disabled=no

The basic telephony configuration should be as follows:

• Add a voip voice port to the /ip telephony voice-port voip for each of the devices you want to call, or want to receive calls from, i.e., (the IP telephony gateway 10.1.1.12 and the Welltech IP telephone 10.5.8.2):

  [admin@Joe] ip telephony voice-port voip> add name=gw remote-address=10.1.1.12
  [admin@Joe] ip telephony voice-port voip> add name=rob remote-address=10.5.8.2
  [admin@Joe] ip telephony voice-port voip> print
  Flags: X - disabled, D - dynamic, R - registered
    #    NAME      AUTODIAL REMOTE-ADDRESS  JITTER-BUFFER PREFERED-CODEC  SIL FAS
    0    gw                 10.1.1.12       100ms         none            no  yes
    1    rob                10.5.8.2        100ms         none            no  yes
  [admin@Joe] ip telephony voice-port voip>
            

  You should have three vioce ports now:

  [admin@Joe] ip telephony voice-port> print
  Flags: X - disabled
    #   NAME                          TYPE           AUTODIAL
    0   linejack1                     linejack
    1   gw                            voip
    2   rob                           voip
  [admin@Joe] ip telephony voice-port>
            

• Add at least one unique number to the /ip telephony numbers for each voice port. This number will be used to call that port:

  [admin@Joe] ip telephony numbers> add dst-pattern=31 voice-port=rob
  [admin@Joe] ip telephony numbers> add dst-pattern=33 voice-port=linejack1
  [admin@Joe] ip telephony numbers> add dst-pattern=1. voice-port=gw prefix=1
  [admin@Joe] ip telephony numbers> print
  Flags: I - invalid, X - disabled, D - dynamic, R - registered
    #     DST-PATTERN             VOICE-PORT              PREFIX
    0     31                          rob                 31
    1     33                          linejack1
    2     1.                          gw                  1
  [admin@Joe] ip telephony numbers>
            

  Here, the dst-pattern=31 is to call the Welltech IP Telephone, if the number 31 is dialed on the dialpad. The dst-pattern=33 is to ring the local telephone, if a call for number 33 is received over the network. Anything starting with digit '1' would be sent over to the IP Telephony gateway.

Making calls from the IP telephone 10.0.0.224:

• To call the IP telephone 10.5.8.2, it is enough to lift the handset and dial the number 31

• To call the PBX extension 13, it is enough to lift the handset and dial the number 13

  After establishing the connection with 13, the voice port monitor shows:

  [admin@Joe] ip telephony voice-port linejack> monitor linejack
                 status: connection
                   port: phone
              direction: port-to-ip
            line-status: unplugged
           phone-number: 13
      remote-party-name: PBX_Line [10.1.1.12]
                  codec: G.723.1-6.3k/hw
               duration: 16s
  
  [admin@Joe] ip telephony voice-port linejack>
            

• Set the regional setting to match our PBX. The mikrotik region will be used in thisn example:

  [admin@voip_gw] ip telephony voice-port linejack> set linejack1 region=mikrotik
  [admin@voip_gw] ip telephony voice-port linejack> print
  Flags: X - disabled
    0   name="linejack1" autodial="" region=mikrotik playback-volume=0
        record-volume=0 ring-cadence="++-++--- ++-++---" agc-on-playback=no
        agc-on-record=no aec=yes aec-tail-length=short aec-nlp-threshold=low
        aec-attenuation-scaling=4 aec-attenuation-boost=0 software-aec=no
        detect-cpt=yes
  
  [admin@voip_gw] ip telephony voice-port linejack>
            

• Add a voip voice port to the /ip telephony voice-port voip for each of the devices you want to call, or want to receive calls from, i.e., (the IP telephone 10.0.0.224 and the Welltech IP telephone 10.5.8.2):

  [admin@voip_gw] ip telephony voice-port voip> add name=joe \
  \... remote-address=10.0.0.224
  [admin@voip_gw] ip telephony voice-port voip> add name=rob \
  \... remote-address=10.5.8.2 prefered-codec=G.723.1-6.3k/hw
  [admin@voip_gw] ip telephony voice-port voip> print
  Flags: X - disabled, D - dynamic, R - registered
    #    NAME      AUTODIAL REMOTE-ADDRESS  JITTER-BUFFER PREFERED-CODEC  SIL FAS
    0    joe                10.0.0.224      100ms         none            no yes
    1    rob                10.5.8.2        100ms         G.723.1-6.3k/hw no yes
  [admin@voip_gw] ip telephony voice-port voip>
            

• Add number records to the /ip telephony numbers, so you are able to make calls:

  [admin@voip_gw] ip telephony numbers> add dst-pattern=31 voice-port=rob prefix=31
  [admin@voip_gw] ip telephony numbers> add dst-pattern=33 voice-port=joe prefix=33
  [admin@voip_gw] ip telephony numbers> add dst-pattern=1. voice-port=linejack1 \
  \... prefix=1
  [admin@voip_gw] ip telephony numbers> print
  Flags: I - invalid, X - disabled, D - dynamic, R - registered
    #     DST-PATTERN             VOICE-PORT              PREFIX
    0     31                          rob                 31
    1     33                          joe                 33
    2     1.                          linejack1           1
  [admin@voip_gw] ip telephony numbers>
            

Making calls through the IP telephony gateway:

• To dial the IP telephone 10.0.0.224 from the office PBX line, the extension number 19 should be dialed, and, after the dial tone has been received, the number 33 should be entered. Thus, the telephone [Joe] is ringed.

  After establishing the voice connection with '33' (the call has been answered), the voice port monitor shows:

  [admin@voip_gw] ip telephony voice-port linejack> monitor linejack1
                 status: connection
                   port: line
              direction: port-to-ip
            line-status: plugged
           phone-number: 33
      remote-party-name: linejack1 [10.0.0.224]
                  codec: G.723.1-6.3k/hw
               duration: 1m46s
  
  [admin@voip_gw] ip telephony voice-port linejack>
            

• To dial the IP telephone 10.5.8.2 from the office PBX line, the extension number 19 should be dialed, and, after the dial tone has been received, the number 31 should be entered.

Making calls from the IP telephone 10.5.8.2:

• Just lift the handset and dial '11', or '13' fo the PBX extensions.

• Dial '33' for [Joe]. The call request will be sent to the gateway 10.1.1.12, where it will be forwarded to [Joe]. If you want to call [Joe] directly, add a phonebook record for it:

  usr/config$ pbook -add name Joe ip 10.0.0.224 e164 33
            

Configuration on the MikroTik side

• G.729a codec MUST be disabled (otherwise connections are not possible at all!!!)

  /ip telephony codec disable G.729A-8k/sw
            

• G.711-ALaw codec should not be used (in some cases there is no sound)

  /ip telephony codec disable "G.711-ALaw-64k/sw G.711-ALaw-64k/hw"
            

• Fast start has to be used (otherwise no ring-back tone and problems with codec negotiation)

  /ip telephony voice-port set cisco fast-start=yes
            

• Telephone number we want to call to must be sent to Cisco, for example

  /ip telephony numbers add destination-pattern=101 voice-port=cisco prefix=101
            

• Telephone number, cisco will call us, must be assigned to some voice port, for example,

  /ip telephony numbers add destination-pattern=098 voice-port=linejack
            

Configuration on the CISCO side:

• IP routing has to be enabled

  ip routing
            

• Default values for fast start can be used:

  voice service pots
      default h323 call start
      exit
  voice service voip
      default h323 call start
      exit
            

• Enable opening of RTP streams:

  voice rtp send-recv
            

• Assign some E.164 number for local telephone, for example, 101 to port 0/0

  dial-peer voice 1 pots
      destination-pattern 101
      port 0/0
      exit
            

• create preferred codec listing:

  voice class codec codec_class_number
      codec preference 1 g711ulaw
      codec preference 2 g723r63
      exit
            

  NOTE: g723r53 codec can be used, too

• Tell, that some foreign E.164 telephone number can be reached by calling to some IP address, for example, 098 by calling to 10.0.0.98

  dial-peer voice 11 voip
      destination-pattern 098
      session target ipv4:10.0.0.98
      voice-class codec codec_class_number
      exit
            

  NOTE: instead of codec class, one specified codec could be specified:

  codec g711ulaw
            

Assume that:

• The IP telephony gateway #1 has IP address 10.0.0.182, and the name of the Voicetronix first line is 'vctx1'.
• The IP telephony gateway #2 has IP address 10.0.0.183, and the name of the Voicetronix first line is 'vctx1'.

The IP telephony configuration should be as follows:

• IP telephony gateway #1 should have:

  /ip telephony voice-port voip
  add name=gw2 remote-address=10.0.0.183
  /ip telephony numbers
  add dst-pattern=1.. voice-port=gw2 prefix=2
  add dst-pattern=2.. voice-port=vctx1 prefix=1
            

• IP telephony gateway #2 should have

  /ip telephony voice-port voip
  add name=gw1 remote-address=10.0.0.182
  /ip telephony numbers
  add dst-pattern=2.. voice-port=vctx1 prefix=1
  add dst-pattern=1.. voice-port=gw1 prefix=2