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##master-page:HomepageReadWritePageTemplate
##master-date:Unknown-Date
#format wiki
#language en
= Cisco VoiP (Voice over IP) =

== Upgrade tool ==
 * Link [[Cisco/VoiP/DSP]] * [[Cisco/ISDN]] * [[VoiP/Delay]]

 * http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080b31514.shtml
 * Bridge CUCM upgrade, upgrade, make DRS backup, intall new version in new vm, recover DRS backup
   * http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cucos/8_0_1/cucos/iptpch7.html#wp1058411

== Call Manager Debug ==
 * On Router MGCP controlled
   * #sh rtp statistics
     {{{
     RTP Statistics info:
     No. CallId Xmit-pkts Xmit-bytes Rcvd-pkts Rcvd-bytes Lost pkts Jitter Latenc
     1 552059 0x486D 0x2D4420 0x486F 0x2D4560 0x0 0x0 0x0
     2 552070 0x20A1 0x1464A0 0x209F 0x146360 0x0 0x0 0x17
     3 552061 0x486C 0x2D4380 0x486C 0x2D4380 0x0 0x0 0x0
     4 552068 0x20A2 0x146540 0x2382 0x163140 0x0 0x0 0x0
     }}}
   * #sh mgcp
     {{{
     MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE
     MGCP call-agent: 10.11.0.111 2427 Initial protocol service is MGCP 0.1
     ...
     MGCP control bound to interface GigabitEthernet0/0.112
     MGCP media bound to interface GigabitEthernet0/0.112
     ...
     MGCP media (RTP) dscp: ef, MGCP signaling dscp: af31
     }}}

== 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.
   1. Phone tftp's to CM option 150 ip to retrieve config file SEP.cnf.xml
      * From wireshark trace,
        a. CTLSEP001F9EABA0F3.tlv (File not found on CM) and then
        a. SEP001F9EABA0F3.cnf.xml (See phone mac ?) (includes, CM ip and code to load e.g. SCCP11.8-5-2S )
        a. English_United_States/tc-sccp.jar
        a. United_States/g3-tones.xml
        a. sccp login on tcp port 2000 to ip from config above.
   1.
  

== CCMX Notes ==
 * Cisco Unified CCX. Cisco Unified CCX Premium.
   * QM and AQM are available only with Cisco Unified CCX Premium.

 *[[http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801a62a8.shtml|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.
   * http://www.cisco.com/en/US/docs/voice_ip_comm/uc_system/UC6.0.1/features_apps/CMmonrec.html
== Next ==
 * Cisco BW calculations, and optimization. http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml#configvoice
   * Packet size, VAD

 * VoiP [[http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_2/ccmsys/a02sycfg.html#wp1038253|Bandwidth]] savings.
   1. Codec: G.711 (64 Kbps) , G.729 ('''8 Kbps''' - Voice)('''24 Kbps''' - with serial overhead) ('''32kbps - on [[http://www.voip-info.org/wiki/view/Bandwidth+consumption|ethernet]]''')
   1. 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.
   1. 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).
 
 * [[http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_2/ccmsys/a02cac.html|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 ==
 * [[http://www.networkers-google.com/nwgoogle/Voice/Others/How%20to%20check%20DSP%20Keepaliv1307703118.html|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 (Voice over IP)

Upgrade tool

Call Manager Debug

  • On Router MGCP controlled
    • #sh rtp statistics
      •      RTP Statistics info:
             No. CallId     Xmit-pkts Xmit-bytes Rcvd-pkts  Rcvd-bytes Lost pkts  Jitter Latenc
             1   552059     0x486D    0x2D4420   0x486F     0x2D4560   0x0        0x0      0x0     
             2   552070     0x20A1    0x1464A0   0x209F     0x146360   0x0        0x0      0x17    
             3   552061     0x486C    0x2D4380   0x486C     0x2D4380   0x0        0x0      0x0     
             4   552068     0x20A2    0x146540   0x2382     0x163140   0x0        0x0      0x0
    • #sh mgcp
      •      MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE
             MGCP call-agent: 10.11.0.111 2427 Initial protocol service is MGCP 0.1
             ...
             MGCP control bound to interface GigabitEthernet0/0.112
             MGCP media bound to interface GigabitEthernet0/0.112
             ...
             MGCP media (RTP) dscp: ef, MGCP signaling dscp: af31

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)