Communication session quality indicator

• US 8,467,308 B2
• Filed: 08/17/2009
• Issued: 06/18/2013
• Est. Priority Date: 10/25/2001
• Status: Expired due to Fees

• Current Assignee: Verizon Patent and Licensing Incorporated (Verizon Communications Inc.)
• Original Assignee: Verizon Business Global LLC (Verizon Communications Inc.)
• Inventors: Johnston, Alan B.
• Primary Examiner(s): Sefcheck, Gregory
• Assistant Examiner(s): SMITH, JOSHUA Y
• Application Number:
  US12/542,473
• Publication Number:
  US 20100040050A1
• Time in Patent Office: 1,401 Days
• Field of Search: None
• US Class Current: 370/252
• CPC Class Codes: H04L 12/6418 Hybrid transport
• CPC Class Codes: H04L 47/10 Flow control; Congestion control
• CPC Class Codes: H04L 47/11 Identifying congestion
• CPC Class Codes: H04L 47/2416 Real-time traffic
• CPC Class Codes: H04L 47/26 using explicit feedback to the source, e.g. choke packets
• CPC Class Codes: H04L 47/38 by adapting coding or compression rate
• CPC Class Codes: H04L 47/431 using padding or de-padding
• CPC Class Codes: H04L 65/1101 Session protocols
• CPC Class Codes: H04L 65/65 Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
• CPC Class Codes: H04L 65/80 Responding to QoS
• CPC Class Codes: H04L 67/14 Session management for real-time applications in data packet communications networks H04L65/1066
• CPC Class Codes: H04L 69/327 in the session layer [OSI layer 5]
• CPC Class Codes: H04L 69/329 in the application layer [OSI layer 7]
• CPC Class Codes: H04L 9/40 Network security protocols
• CPC Class Codes: H04M 3/2227 Quality of service monitoring
• CPC Class Codes: H04M 7/006 Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP) , including next generation networks with a packet-switched transport layer H04L65/00 takes precedence; aspects not specific to the type of network H04M3/00; special services in those networks H04M3/42

Abstract

An approach for providing a quality indicator in support of a communication session between a near end station and a far end station over a data network. The quality of the communication session in a direction of the near end station sending to the far end station is determined. A message containing statistics associated with the communication session is transmitted according to a prescribed protocol to the near end station to notify the near end station of the quality of the communication session. The prescribed protocol supports real-time data exchange. The present invention has particular applicability to SIP (Session Initiation Protocol) IP (Internet Protocol) telephony services.

Claims

1. A method comprising:

generating a plurality of packets for transmission over a first voice path of a voice call with an end station, wherein the first voice path is in a direction to the end station, the voice call having a second voice path in a direction from the end station;

receiving a report packet from the end station indicating quality of the first voice path, wherein the report packet includes a field specifying number of lost packets;

generating a notification for presentation to a user engaged in the voice call, the notification specifying one of a plurality of levels corresponding to the quality of the first voice path based on the report packet; and

selectively initiating termination of the voice call based on the generated notification,

wherein the report packet further includes a field specifying a timestamp corresponding to a time when the report packet was transmitted from the end station, a field specifying a fraction of packets lost from the end station since a previous time a report packet was sent, and a field specifying an estimate of a statistical variance of a packet interarrival time.

2. A method according to claim 1, wherein the voice call is established according to a Real-Time Transport Protocol (RTP) and the report packet is generated according to a Real-Time Transport Control Protocol (RTCP).

3. A method according to claim 1, wherein the notification is either visual, audio, or a combination thereof.

4. A method according to claim 3, wherein the visual notification is represented by either a LED (Light Emitting Diode)-type display, or a pop-up menu.

5. A method according to claim 1, wherein the voice call is established according to a Session Initiation Protocol (SIP).

6. A method according to claim 1, wherein the termination is initiated in response to an input by the user.

7. An apparatus comprising:

a processor configured to generate a plurality of packets for transmission over a first voice path of a voice call with an end station, wherein the first voice path is in a direction to the end station, the voice call having a second voice path in a direction from the end station; and

a communication interface coupled to the processor and configured to receive a report packet from the end station indicating quality of the first voice path, the report packet including a field specifying number of lost packets; and

wherein the processor is further configured to generate a notification for presentation to a user engaged in the voice call, the notification specifying one of a plurality of levels corresponding to the quality of the first voice path based on the report packet, the processor being further configured to selectively initiate termination of the voice call based on the generated notification,

wherein the report packet further includes a field specifying a timestamp corresponding to a time when the report packet was transmitted from the end station, a field specifying a fraction of packets lost from the end station since a previous time a report packet was sent, and a field specifying an estimate of a statistical variance of a packet interarrival time.

8. An apparatus according to claim 7, wherein the voice call is established according to a Real-Time Transport Protocol (RTP) and the report packet is generated according to a Real-Time Transport Control Protocol (RTCP).

9. An apparatus according to claim 7, wherein the notification is either visual, audio, or a combination thereof.

10. An apparatus according to claim 9, wherein the visual notification is represented by either a LED (Light Emitting Diode)-type display, or a pop-up menu.

11. An apparatus according to claim 7, wherein the voice call is established according to a Session Initiation Protocol (SIP).

12. An apparatus according to claim 7, further comprising:

a user interface configured to receive an input by the user, wherein the termination is initiated in response to the input by the user.

13. A method comprising:

receiving a plurality of packets over a first voice path of a voice call with an end station, wherein the first voice path is in a direction from the end station, the voice call having a second voice path in a direction to the end station;

generating a report packet indicating quality of the first voice path, wherein the report packet includes a field specifying number of lost packets;

initiating transmission of the report packet to the end station; and

selectively receiving a request from the end station to terminate the voice call based on the transmitted report packet,

wherein the report packet further includes a field specifying a timestamp corresponding to a time when the report packet was transmitted to the end station, a field specifying a fraction of packets lost since a previous time a report packet was sent, and a field specifying an estimate of a statistical variance of a packet interarrival time.

14. A method according to claim 13, wherein the voice call is established according to a Real-Time Transport Protocol (RTP) and the report packet is generated according to a Real-Time Transport Control Protocol (RTCP).

15. An apparatus comprising:

a communication interface configured to receive a plurality of packets over a first voice path of a voice call with an end station, wherein the first voice path is in a direction from the end station, the voice call having a second voice path in a direction to the end station; and

a processor coupled to the communication interface and configured to generate a report packet indicating quality of the first voice path, the report packet including a field specifying number of lost packets, the processor being further configured to initiate transmission of the report packet to the end station,

wherein the communication interface is further configured to selectively receive a request from the end station to terminate the voice call based on the transmitted report packet, and

wherein the report packet further includes a field specifying a timestamp corresponding to a time when the report packet was transmitted to the end station, a field specifying a fraction of packets lost since a previous time a report packet was sent, and a field specifying an estimate of a statistical variance of a packet interarrival time.

16. An apparatus according to claim 15, wherein the voice call is established according to a Real-Time Transport Protocol (RTP) and the report packet is generated according to a Real-Time Transport Control Protocol (RTCP).