Basic Cisco IP Telephony call flow using SIP and SCCP

CUCM Signalling and Media Paths – Basic IP Telephony call flow using SCCP and SIP Protocol

I am sure most of you are already working in IP Telephony for a long time and by now you already know the signalling and media path used by CUCM when the phone uses SIP or SCCP Protocol. This post is dedicated to the upcoming IPT Engineers who has just started learning Cisco VoIP. In this post we will be talking about signalling and media path taken when a call is made.

Let’s assume there are two users i.e., User A and User B. User A will make a call to User B using SCCP and SIP Protocol and lets see how the call flows.

SCCP and SIP Call flows:

  1. User A goes off-hook (picks up the handset). As soon as the user goes off hook, a message is sent to Cisco Unified Communications Manager (CUCM) letting it know that one of the device (IP Phone) has gone off-hook.
  2. CUCM will respond to the event with a message which tells the device to play a dial tone that is stored in the flash memory of the IP Phone.
  3. User A hears the dial tone and he starts dialing the phone number of User B.
  4. In case if the IP Phone is configured to use SCCP Protocol, the digits are sent to CUCM as they are pressed i.e., Digit by Digit. For example: if the User B’s number is 1000, IP Phone will first send 1, then it sends 0, then it sends 0 and again it sends 0.
  5. In case if the IP Phone is configured to use SIP Protocol, the digits are sent to CUCM in one single message i.e, enbloc signalling. For example: if the User B’s number is 1000, IP Phone will send 1000 in single message.
  6. Once the number is dialed, CUCM will perform digit analysis against the dialed digits. If a valid match is found, CUCM will route the call as per it’s configuration whereas if it does not find a match, a re-order tone will be sent to User A.
  7. CUCM will signal the calling party to initiate ringback so the User A will hear a ring back tone. CUCM will also signal the call to the destination phone and a ringdown tone is played.
  8. The Phones will be provided information such as Calling and Called Party Name and Number. User A’s Phone will show the destination device name and number and User B’s Phone will show the calling party name and number.
  9. Once User B answers the call, CUCM will send a message to the devices letting them know the IPv4 socket i.e., IP Address and Port number in which they should communicate for the duration of the call. The RTP media path opens directly between User A’s phone and User B’s phone.  The Cisco IP Phones require no further communication with CUCM until either phone invokes a feature, such as call conferencing, call termination or call transfer. The call will continue to work even the CUCM goes down.

Hope this helps!

3 thoughts on “Basic Cisco IP Telephony call flow using SIP and SCCP”

  1. Mike says:

    Good read.

    1. Administrator says:

      Thanks Mike! Keep following and keep sharing.

  2. eddie says:

    It great article!!
    By the way ~
    SIP Protocol have two type that A and B
    I think maybe you can explanation what they different

    thanks your article help me a lots!

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