Method and apparatus for management of voice-over IP communications

• US 20050052996A1
• Filed: 09/09/2003
• Published: 03/10/2005
• Est. Priority Date: 09/09/2003
• Status: Abandoned Application

• Current Assignee: Lucent Technologies, Inc. (Nokia Corporation)
• Original Assignee: Lucent Technologies, Inc. (Nokia Corporation)
• Inventors: Kim, Eunyoung, Wehr, Larry A., Uzunalioglu, Huseyin, Houck, David J.
• Application Number:
  US10/657,864
• Publication Number:
  US 20050052996A1
• Time in Patent Office: Days
• US Class Current: 370/230
• CPC Class Codes: H04L 47/11 Identifying congestion
• CPC Class Codes: H04L 47/12 Avoiding congestion; Recovering from congestion
• CPC Class Codes: H04L 47/29 using a combination of thresholds
• CPC Class Codes: H04L 47/365 Dynamic adaptation of the packet size
• CPC Class Codes: H04L 47/38 by adapting coding or compression rate
• CPC Class Codes: H04L 47/70 Admission control; Resource allocation
• CPC Class Codes: H04L 47/745 Reaction in network
• CPC Class Codes: H04L 47/762 triggered by the network
• CPC Class Codes: H04L 47/801 Real time traffic
• CPC Class Codes: H04L 47/822 Collecting or measuring resource availability data
• CPC Class Codes: H04L 47/825 Involving tunnels, e.g. MPLS
• CPC Class Codes: H04L 47/826 Involving periods of time
• CPC Class Codes: H04L 65/1026 at the edge
• CPC Class Codes: H04L 65/1036 at the edge
• CPC Class Codes: H04L 65/1043 Gateway controllers, e.g. media gateway control protocol [MGCP] controllers
• CPC Class Codes: H04L 65/1069 Session establishment or de-establishment
• CPC Class Codes: H04L 65/1101 Session protocols
• CPC Class Codes: H04L 65/752 adapting media to network capabilities
• CPC Class Codes: H04L 65/80 Responding to QoS

Abstract

A method and apparatus for analyzing current voice call quality in an IP network before allowing a new voice call to enter therein. Circuitry in each gateway between a PSTN and the IP network obtains information from the IP network regarding the quality of voice call traffic being transmitted from one gateway to another gateway. A parameter is calculated by the circuitry based on this information. This parameter is compared to predetermined thresholds to guarantee acceptable quality for a new voice call that is attempting to enter the IP network. The circuitry includes three circuits. A first circuit passes voice call data to the IP network. A second circuit polls the IP network about traffic information transmitted therein. A third circuit processes the polled information to determine whether the voice call data is to be accepted by the IP network.

Claims

1. A method, comprising the steps of:

(a) obtaining information relevant to the quality of service of voice calls being transmitted from a first location to a second location via an IP network;

(b) calculating a parameter based on said information; and

(c) accepting a new call into the IP network in the case of said parameter not exceeding an upper threshold.

2. The method of claim 1 wherein said new call is accepted into the IP network at a reduced bandwidth in the case of said parameter exceeding a lower threshold.

3. The method of claim 1 where said new call is not accepted into the IP network in the case of said parameter exceeding the upper threshold.

4. The method of claim 1 wherein the information obtained is a number of lost packets, late packets and received packets (collectively defined as “

sent”

packets) transmitted from said first location to said second location in the IP network.

5. The method of claim 1 wherein the information obtained is a delay of received packets transmitted from said first location to said second location in the IP network.

6. The method of claim 1 wherein the information obtained is a delay variation of received packets transmitted from said first location to said second location in the IP network.

7. The method of claim 1 wherein the information is obtained on a periodic basis.

8. The method of claim 1 wherein the information is obtained on an exception basis using an immediate report.

9. The method of claim 1 wherein the parameter is identified as a packet lost ratio (PLR).

10. The method of claim 9 wherein PLR is defined as

11. The method of claim 2 wherein bandwidth is reduced for a newly accepted call by selecting a first encoder to encode the new voice call information in a bandwidth that is smaller than bandwidths of other calls accepted in the network that are encoded by a second encoder.

12. The method of claim 2 wherein the bandwidth of a newly accepted call is reduced by increasing the packet size (voice sample) for said newly accepted voice call.

13. The method of claim 2 wherein the bandwidth of a newly accepted call is reduced by activating the characteristic of silence suppression for said newly accepted voice call.

14. Apparatus comprising a gateway for interfacing voice call data from a public switch telephone network to an internet protocol network;

said gateway further comprising;

a first circuit for passing said voice call data to the internet protocol network;

a second circuit for polling the internet protocol network about traffic information transmitted therein; and

a third circuit for processing the polled information to determine whether the voice call data is to be accepted by the internet protocol network.

15. The apparatus of claim 14 wherein said first circuit further comprises one or more Ethernet cards that are connected to the internet protocol network.

16. The apparatus of claim 14 wherein said second circuit is at least one strongarm card.

17. The apparatus of claim 16 wherein the strongarm card is connected to the Ethernet card via a host CPU circuit.

18. The apparatus of claim 14 wherein the third circuit compares a parameter based on the polled information to a plurality of thresholds.

19. The apparatus of claim 18 wherein the parameter is a packet loss ratio defined as

20. The apparatus of claim 19 wherein the third circuit compares the packet loss ratio to a lower threshold and if the packet loss ratio is less than the lower threshold, a new voice call is accepted into the internet protocol network.

21. The apparatus of claim 19 wherein the third circuit compares the packet loss ratio to the lower threshold and an upper threshold, and if lower threshold<

packet loss ratio<

upper threshold, a new voice call is accepted into the internet protocol network at a reduced bandwidth.

22. The apparatus of claim 19 wherein the third circuit compares the packet loss ratio to the upper threshold, and if the packet loss ratio is greater than the upper threshold, a new voice call is blocked from entering the internet protocol network.