• SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only.
• Cisco Unified CME 3.4 and later versions support Media Flow-through mode only; enabling SIP-to-SIP calls is required before you can successfully make SIP-to-SIP calls.
• Media Flow-around configured with the media flow-around command is not supported by Cisco Unified CME with SIP phones.

• Allow one SIP call leg to another SIP call leg
• Configure router as a registrar server. To allow SIP UA/Phone to register their DN and IP
  • Max
    • Default is 3600 and recommended value is 600
    • Cisco Unified CME does not maintain a persistent database of registration entries across reloads. Because SIP phones do not use a keepalive functionality, the SIP phones must register again. To decrease the amount of time after which the SIP phones register again, we recommend that you change the expiry.
  • Min
    • Ensure that the registration expiration timeout is set to a value smaller than the TCP connection aging timeout to avoid disconnection from the TCP.
    • Default is 60
• Specify the source interface for all sip packets originated from this router
  • If you don't configure this, ccme may use another IP to initiate connection to the ipphone which will cause some of the older ip phone registration to fail registration

voice service voip
  allow-connections sip to sip
  !
  sip
    registrar server expires max <secs> min <secs>
    bind all source-interface vlan 400

• Mode cme - configure router as a sip proxy to handle call routing.
• Max-dn is default to 0.
  • Do not set this number higher than what's required as it consumes memory
• Max-pool is default to 0.
  • Again do not set this number higher than what's required as it consumes memory
  • This number actually restrict the max voice register pool number you can configure i.e if configured as 10 then you can not configure a voice register pool 11
• Specify the IP
• Create phone configuration files

voice register global
  mode cme
  source-address <cme-ip> [port <port-num>]
  max-dn <number>
  max-pool <number>
  !
  create profile

• Use a loopback interface to SIP

voice service voip
sip
  bind control source-interface loop 0

• Shows SIP telephony service

CCME# show voice register global