##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]]  * [[SIP]]

 * 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