US 7,475,003 B1
Method and apparatus for initiating call analysis using an internet protocol phone
Deerendra Madhusudhana, Karnataka (India); and Girish M. Ganeshmurthy, Karnataka (India)
Assigned to Cisco Technology, Inc., San Jose, Calif. (US)
Filed on Oct. 09, 2003, as Appl. No. 10/683,954.
Int. Cl. G06F 15/173 (2006.01)
U.S. Cl. 703—223  [709/217; 709/219; 370/356] 22 Claims
OG exemplary drawing
 
1. A method for initiating analysis of a call from an Internet Protocol (IP) phone, the method comprising computer-implemented steps of:
accessing, by the phone, configuration information associated with the phone, wherein the configuration information includes information about an associated network management system;
automatically initiating, from the phone, transmitting an alert to the network management system, wherein the alert informs the network management system about the call and requests the analysis;
determining, by the phone, an issue with the call that warrants analysis of the call by performing steps selected from the group consisting of:
(a) determining that an elapsed time from the phone going off hook to receiving a message at the phone that instructs the phone to play a dial tone exceeds a particular value;
(b) transmitting to a call manager a representation of a phone number that is associated with a called party, and determining that a message was not received from the call manager in response to the representation;
(c) waiting for RTP packets from a called endpoint, and determining that a particular time interval has elapsed before receiving an RTP packet from the called endpoint;
(d) determining that a play-out time interval that is associated with a dejitter buffer that is associated with the phone is greater than a particular value;
(e) recording a first number of packets that are dropped before reaching the phone, by using a previous packet sequence number and a current packet sequence number, recording a second number of packets that are dropped by a dejitter buffer that is associated with the phone, by using a previous packet sequence number and a current packet sequence number, and determining that a product of a sum of the first number of packets that are dropped before reaching the phone and the second number of packets that are dropped by the dejitter buffer, and a packetization delay that is associated with a codec that is associated with the call, is greater than a particular value;
(f) determining that RTP packets are not received continuously by the phone for a period greater than a particular value; and
(g) determining that a ratio of total packets lost before reaching the phone divided by total packets received at the phone is greater than a particular value.