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== Call Manager Debug ==
 * On Router MGCP controlled
   * #sh rtp statistics

Cisco VoiP (Voice over IP)

Call Manager Debug

  • On Router MGCP controlled
    • #sh rtp statistics

Cisco phones

7911

  • Configures for sccp. (Not using SIP yet)
  • Boot process
    1. DHCP gives phone ip, and option 150 gives it the CM ip's.
    2. Phone tftp's to CM option 150 ip to retrieve config file SEP.cnf.xml
      • From wireshark trace,
        1. CTLSEP001F9EABA0F3.tlv (File not found on CM) and then
        2. SEP001F9EABA0F3.cnf.xml (See phone mac ?) (includes, CM ip and code to load e.g. SCCP11.8-5-2S )
        3. English_United_States/tc-sccp.jar
        4. United_States/g3-tones.xml
        5. sccp login on tcp port 2000 to ip from config above.

CCMX Notes

  • Cisco Unified CCX. Cisco Unified CCX Premium.
    • QM and AQM are available only with Cisco Unified CCX Premium.
  • Note: Voice monitoring and recording are supported only on IPCC Express Enhanced and Premium Edition, not on Standard Edition.

  • The VoIP silent monitoring server must be on the same VLAN as the agent phones and requires an available SPAN port.
  • Cisco VoIP call recording with CallRex Call Recording™ software can be achieved through either packet-sniffing via port mirroring or with forked audio using the Built-in Bridge in selected Cisco IP-based telephones.

  • Call recording can be achieved using the Built-in Bridge (BIB) option on selected Cisco IP phones.

Next

  • Cisco BW calculations, and optimization. http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml#configvoice

    • Packet size, VAD
  • VoiP Bandwidth savings.

    1. Codec: G.711 (64 Kbps) , G.729 (8 Kbps - Voice)(24 Kbps - with serial overhead) (32kbps - on ethernet)

    2. Changing Voice Payload Sizes
      • G.729 call with voice payload size of 20 bytes (20 ms):= 24 Kbps 40 bytes (40 ms):= 16 Kbps
        • Note: L2 headers are not considered in this calculation.
    3. RTP Header-Compression or Compressed RTP (cRTP)
      • Compressed Real-Time Protocol (cRTP) reduces the IP/UDP/RTP headers to 2or 4bytes (cRTP is not available over Ethernet).
  • Call Admission Control

    • Locations, for systems with centralized call processing. Bandwidth (kbps)
    • Regions define the type of compression (G.711, G.723, or G.729) that is used on the link, and locations define the amount of available bandwidth for the link.
    • You assign each device in the system to both a region (by means of a device pool) and a location.

DSP

  • check DSP Keepalive status in Cisco Voice gateways.

    • Voice-GW#test voice driver
    • # sh voice dsp
    • # show call resource voice stats
      • If above gives 0, you have not enabled dsp services dspfarm under the voice-card

  • Setup BRI at remote site for breakout using central callmanager, H.323
    •       !
            voice-card 0
              dspfarm
              dsp services dspfarm
            !
            voice call carrier capacity active
            voice rtp send-recv
            !
            voice service voip 
              h323
            !
            interface FastEthernet0/0.x
              description VOICE-Vlan
              ip address XX
              h323-gateway voip interface
              h323-gateway voip bind srcaddr XX
            !
            interface BRI1/0
              description ISDN-VoicePort
              no ip address
              isdn switch-type basic-net3
              isdn point-to-point-setup
              isdn incoming-voice voice
              isdn send-alerting
              isdn sending-complete
            !
            voice-port 1/0
              compand-type a-law
            !
            voice-port 1/1
              compand-type a-law
            !
            !
            dial-peer voice 20 voip
              destination-pattern 01132699..
              voice-class codec 1
              session target ipv4:10.110.0.102
              incoming called-number 01132699..
              dtmf-relay h245-alphanumeric
              no vad
            !
            dial-peer voice 3 pots
              destination-pattern .T
              progress_ind alert enable 8
              incoming called-number 9...
              direct-inward-dial
              port 0/1
              forward-digits all
           !

...


CategoryCisco

Cisco/VoiP (last edited 2017-11-05 10:04:36 by PieterSmit)