====== CCME SIP ======
  * 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. 

===== Configuration =====

==== Voice Service VOIP ====
  * 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
<code>
voice service voip
  allow-connections sip to sip
  !
  sip
    registrar server expires max <secs> min <secs>
    bind all source-interface vlan 400
</code>

==== Telephony Service ====
  * 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
<code>
voice register global
  mode cme
  source-address <cme-ip> [port <port-num>]
  max-dn <number>
  max-pool <number>
  !
  create profile
</code>

===== Loopback =====
  * Use a loopback interface to SIP
<code>
voice service voip
sip
  bind control source-interface loop 0
</code>

===== Debug =====
  * Shows SIP telephony service
<code>
CCME# show voice register global
</code>