Automated session initiation protocol (SIP) device

• US 8,873,405 B2
• Filed: 12/15/2006
• Issued: 10/28/2014
• Est. Priority Date: 12/15/2006
• Status: Active Grant

• Current Assignee: Verizon Patent and Licensing Incorporated (Verizon Communications Inc.)
• Original Assignee: Verizon Patent and Licensing Incorporated (Verizon Communications Inc.)
• Inventors: Newman, Craig E.
• Primary Examiner(s): HSU, ALPUS
• Application Number:
  US11/611,582
• Publication Number:
  US 20080144606A1
• Time in Patent Office: 2,874 Days
• Field of Search: 370/242, 370/248, 370/249, 370/356, 370/401, 370/352, 370/250, 370/271, 370/395.52, 714/712, 714/716, 714/717, 379/10.01, 379/27.04, 379/88.17, 379/207.02, 379/418, 709/203, 709/224, 709/227
• US Class Current: 370/250
• CPC Class Codes: H04L 43/50 Testing arrangements
• CPC Class Codes: H04L 65/1059 End-user terminal functionalities specially adapted for real-time communication
• CPC Class Codes: H04L 65/1104 Session initiation protocol [SIP]
• CPC Class Codes: H04M 3/241 for stored program controlled exchanges

Abstract

A device includes a Session Initiation Protocol (SIP) phone, an audio interface for receiving or transmitting audio information with the SIP phone to test a SIP-based network, and a controller for controlling the SIP phone and the audio interface.

Claims

1. A method comprising:

monitoring, by a device, for an occurrence of a ringing event at an automated Session Initiation Protocol (SIP) device (ASD),

the ringing event being associated with a call,

the call occurring after communication is enabled between the ASD and a SIP-based network,

the ASD transmitting a notification when the ringing event occurs;

instructing, by the device and after the ASD has transmitted the notification, the ASD to answer the call;

instructing, by the device and after the ASD has answered the call, a voice server to generate a tone associated with the ASD to test performance of the SIP-based network; and

causing, by the device, the performance of the SIP-based network to be tested using the generated tone associated with the ASD.

2. The method of claim 1, further comprising:

activating or deactivating relays of the ASD based on;

the communication being enabled between the ASD and the SIP-based network, and

transmitting one or more instructions, to the ASD, relating to testing the performance of the SIP-based network.

3. The method of claim 2, where activating or deactivating the relays of the ASD comprises at least one of:

activating a hook-switch of the ASD to answer the call;

disconnecting a SIP phone of the ASD;

dialing telephone numbers of varying lengths via the SIP phone of the ASD;

pressing function keys to invoke a feature via the SIP phone of the ASD;

or

controlling power supplied to the SIP phone of the ASD in order to reset the SIP phone and to cause the SIP phone to receive new configuration information.

4. The method of claim 1, further comprising:

monitoring the ASD for an occurrence of at least one of a call waiting event at the ASD or a message waiting indicator event at the ASD; and

receiving another notification, from the ASD, when the at least one of the call waiting event or the message waiting indicator event occurs.

5. The method of claim 4, where the message waiting indicator event includes a visible light, and

where monitoring for the occurrence of the at least one of the call waiting event or the message waiting indicator event comprises:

monitoring a light source of the visible light to determine whether the message waiting indicator event has occurred.

6. The method of claim 1, where monitoring for the occurrence of the ringing event comprises:

monitoring an audible ringing provided by a speaker of the ASD for a predetermined ring frequency.

7. A device comprising:

a processor to;

monitor an occurrence of a ringing event for a Session Initiation Protocol (SIP) phone associated with the device,

provide, to another device, a notification when the ringing event occurs,

receive, from the other device and based on providing the notification, an instruction to answer a call associated with the ringing event,

cause the SIP phone to answer the call, and

generate analog audio signals based on Real Time Protocol (RTP) packets associated with the call answered by the SIP phone; and

an audio interface to;

transmit the analog audio signals to test performance of a SIP-based network associated with the SIP phone.

8. The device of claim 7, where the device further comprises the SIP phone,

where the SIP phone provides SIP-based communications, and

where the SIP phone includes at least one of a keypad, function keys, a hook switch, a power supply, a speaker, or a microphone.

9. The device of claim 7, where the audio interface comprises:

a transmitter that receives the analog audio signals from the SIP phone, converts the analog audio signals into electrical signals representative of the analog audio signals, and outputs the electrical signals for testing;

a receiver that receives electrical signals representative of audio information; and

a microphone interface that converts the electrical signals, of the receiver, into audio information, and outputs the converted audio information via the SIP phone.

10. The device of claim 7, further comprising:

a relay interface including relays that are activated or deactivated based on instructions received, by the device, from the other device; and

a ring detector to monitor, with the processor, the occurrence of the ringing event for the SIP phone, and to provide the notification, to the other device, when the ringing event occurs.

11. The device of claim 10, where the relay interface, based on the instructions, at least one of:

controls a hook-switch;

dials telephone numbers of varying lengths via the SIP phone;

activates function keys;

or

resets the SIP phone to permit the SIP phone to receive new configuration information.

12. The device of claim 10, where the ring detector:

connects to a speaker, provided within the SIP phone, or the audio interface, and

monitors an audible ringing provided by the speaker of the SIP phone, or the audio interface for a predetermined ring frequency.

13. A method comprising:

translating, by a server, a script command into a communication for a Session Initiation Protocol (SIP) device,

the script command relating to testing a SIP-based network;

mapping, by the server, an Internet Protocol (IP) address and a port location of the SIP device based on the translated script command;

controlling, by the server, the SIP device based on;

the mapped IP address,

the mapped port location, and

the translated script command; and

monitoring, by the server, an event from the SIP device controlled based on;

the mapped IP address,

the mapped IP port location, and

the translated script command,

the script command including a particular script command that creates a test case for testing the SIP-based network.

14. The method of claim 13, where the event includes a ringing event.

15. The method of claim 13, where mapping the IP address and the port location comprises:

providing a memory that stores information enabling reference to a specific IP address and port combination, associated with each of a plurality of SIP devices, for communication with each of the plurality of devices,

where the plurality of SIP devices include the SIP device; and

utilizing the memory when translating the script command.

16. A device comprising:

a script translator to translate a script command into a communication for a Session Initiation Protocol (SIP) device,

the script command relating to testing a SIP-based network;

a memory to map an Internet Protocol (IP) address and a port location of the SIP device based on the translated script command; and

a processor to;

control the SIP device based on;

the mapped IP address,

the mapped port location, and

the translated script command, and

monitor an event from the SIP device controlled based on;

the mapped IP address,

the mapped port location, and

the translated script command,

the script command including a script command that creates a test case to test the SIP-based network.

17. The device of claim 16, where the event includes a ringing event.

18. The device of claim 16, where the processor utilizes the translated script command to provide, to the SIP device, information that relates to testing the SIP-based network.

19. A method comprising:

defining, by a first server, an originating automated Session Initiation Protocol (SIP) device (ASD) and a terminating ASD;

utilizing, by the first server, a database to map an Internet Protocol (IP) address and a port location to the originating ASD and map an IP address and a port location to the terminating ASD;

verifying, by the first server, that the originating ASD is providing a dial tone;

instructing, by the first server and when the originating ASD is providing a dial tone, the originating ASD to dial a telephone number of the terminating ASD to initiate a call;

monitoring, by the first server, for a ringing event at the terminating ASD after instructing the originating ASD to dial the telephone of the terminating ASD;

instructing, by the first server, the terminating ASD to answer the call;

instructing, by the first server, a second server to generate a tone associated with the originating ASD to test performance of a SIP-based network,

the SIP-based network being associated with the originating ASD and the terminating ASD; and

causing, by the first server, the performance of the SIP-based network to be tested by causing a result of the generated tone to be verified.

20. The method of claim 19, further comprising:

instructing the second server to generate a return tone associated with the terminating ASD; and

verifying a result of the generated return tone.

21. The method of claim 20, further comprising:

causing one or more testing functions, of the SIP-based network, to be performed based on at least one of;

verifying the result of the generated tone, or

verifying the result of the generated return tone.

22. The method of claim 21, where causing the one or more testing functions, of the SIP-based network, to be performed includes:

performing a testing function associated with a voice quality associated with at least one of the generated tone or the generated return tone.

23. The method of claim 19, where the ringing event is associated with the call, and

where monitoring for the ringing event includes monitoring for the ringing event, associated with the call, at the terminating ASD.