This document intends to help the beginners to understand the basics of IP Telephony. It explains the working concept of the SCCP & SIP Phones and the Dial rules, digit forwarding methods to initiate a call.
IP Phone Registration Process
As a prerequisite understand the IP Phone Registration Process. Refer the below link to understand the SCCP & SIP Phone Registration Process.
It is nothing but Keypad Markup Language (KPML). Call came from an SIP Phone using KPML, CUCM receives, interprets and analyzes the dial plan on a digit-by-digit basis. Cisco SIP Phones that support KPML use digit-by-digit dialing by default. KPML is not supported on older SIP Phones. Type A phones.
How SCCP & SIP Phone Works?
Cisco IP Phones using SCCP, TCP port 2000 report every user input even to CUCM immediately.
As soon as the user goes off-hook, a signaling message is sent from the phone to the CUCM server with which it is registered. A user goes off-hook and then dials extension 1000. Each event is reported to CUCM in real time.
All the call progress feedback provided by CUCM to the end user on the Cisco IP Phone, Screen display messages showing calling or called party, dial tone, secondary dial tone, ring back, reorder tone and so on is initiated by an SCCP message sent from CUCM to the Cisco IP Phone.
Skinny Client Control Protocol (SCCP) is a stimulus/response protocol where the endpoint sends user input (stimulus) and expects some type of response from the server instructing the endpoint about what to do.
Note: - It is not possible to configure dial plan information (dial rules) on Cisco IP Phones using SCCP. All dial plan functionality is contained in the CUCM cluster with an SCCP phone.
A user dialing an international pattern that is denied by the end user's CoS deployed in CUCM will result in the reorder tone ie busy signal that is played to the calling party letting the party know that the "Call could not be completed as dialed".
If calls to the 976 area code are denied based on the calling party's configured CoS, a reorder tone is sent to the calling party phone as soon as the user dials 91976.
SIP IP Phones:
Type A SIP Phones: Cisco Unified IP Phone models 7905, 7912, 7940 and 7960
Type A SIP Phones support SIP dial rules, which are configured in CUCM and downloaded to the IP Phone at boot time. It does not support KPML. SIP dial rules will enable dial tone and traditional phone functionality on CUCM.
Type A SIP Phones: No Dial rules
Type A Cisco IP Phones using SIP firmware without SIP dial rules do not deliver a dial tone to the calling party when the calling party goes off-hook with the handset or speaker phone. All digits are sent to CUCM enbloc after the user completes the dialing and press the Dial softkey.
Figure shows, User making a call to extension 1000. The user dials 1000 followed by pressing the Dial soft key. The Cisco IP Phone sends a SIP INVITE message to CUCM with all dialed digits (enbloc). CUCM performs digit analysis and provides a call progress message to the IP Phone.
Type A SIP Phones: Dial rules
SIP dial rules allow SIP Phones to emulate the functionality of an SCCP Phone with dial tone and digit-by-digit pattern analysis. When SIP dial rules are leveraged, a user receives dial tone when going off-hook. This functionality is different than the default functionality on Cisco Type A phones converted to SIP, where there are no local dial rules.
Dialed digits are processed against the local SIP dial rules in real time. If a user dials a phone number that is rejected by the local SIP dial rules, for example pay dialing beginning with 9-1900 the call is dropped without being forwarded to CUCM. SIP dial rules can help minimized overhead bandwidth consumption and CUCM processor overhead.
A SIP INVITE message with enbloc signaling is sent from the Cisco SIP IP Phone to CUCM when the SIP dial rule of the Cisco IP Phone recognizes and permits the dialed pattern. End user do not need to press the Dial key like they had to on Cisco SIP type A phones. SIP dial rules allow Type A Phones to emulate SCCP and traditional phone systems while also providing processing and signaling overhead benefits.
Figure illustrates a phone configured to recognize all four-digit patterns beginning with a leading digit of 1. This pattern has an associated timeout value of 0. All user input actions matching the pattern will trigger the sending of the SIP INVITE message to CUCM immediately, without requiring the user to press the Dial key.
Type B SIP Phones: Cisco IP Phone models 79x1, 79x2, 79x5, 7970 and 7960
Type B and new model phones supports KPML and SIP dial rules. KPML is turned on by default on all Type B phone models 79x1, 79x2, 79x5, 7970 and7960
Type B SIP Phones: No Dial Rules
Cisco Type B SIP IP Phones offer functionality based on the KPML standard to report user activities. Each user input event (dialed digit or softkey/button) generates a KPML message to CUCM. This mode of operation emulates a similar end-user experience to that of phones using SCCP.
Type B SIP Phones: Dial Rules
Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules.
Figure illustrates the enbloc signaling used with SIP dial rules.
Note: Cisco SIP Phones that support KPML use digit-by-digit dialing by default. KPML is not supported on older Cisco Type A SIP Phones. Type A phones only support enbloc dialing when using SIP firmware.