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
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
- #sh rtp statistics
Cisco phones
7911
- Configures for sccp. (Not using SIP yet)
- Boot process
- DHCP gives phone ip, and option 150 gives it the CM ip's.
- Phone tftp's to CM option 150 ip to retrieve config file SEP.cnf.xml
- From wireshark trace,
- CTLSEP001F9EABA0F3.tlv (File not found on CM) and then
- SEP001F9EABA0F3.cnf.xml (See phone mac ?) (includes, CM ip and code to load e.g. SCCP11.8-5-2S )
- English_United_States/tc-sccp.jar
- United_States/g3-tones.xml
- sccp login on tcp port 2000 to ip from config above.
- From wireshark trace,
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.
Codec: G.711 (64 Kbps) , G.729 (8 Kbps - Voice)(24 Kbps - with serial overhead) (32kbps - on ethernet)
- 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.
- G.729 call with voice payload size of 20 bytes (20 ms):= 24 Kbps 40 bytes (40 ms):= 16 Kbps
- 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).
- 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 !
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