Dynamically adapting the transmission rate of packets in real-time VoIP communications to the available bandwidth

• US 7,460,480 B2
• Filed: 03/11/2005
• Issued: 12/02/2008
• Est. Priority Date: 03/11/2004
• Status: Active Grant

• Current Assignee: Upland Software, Inc.
• Original Assignee: I2Telecom International Inc.
• Inventors: Awais, Ali
• Primary Examiner(s): Moe, Aung S
• Assistant Examiner(s): Alia, Curtis A
• Application Number:
  US11/078,059
• Publication Number:
  US 20050201414A1
• Time in Patent Office: 1,362 Days
• Field of Search: 370/235, 370/236, 370/252, 370352-356, 370400-401, 370465-467, 379/93.01, 379/93.08
• US Class Current: 370/236
• CPC Class Codes: H04L 41/0896 Bandwidth or capacity management, i.e. automatically increasing or decreasing capacities flow or congestion control using dynamic resource allocation, e.g. in-call renegotiation, H04L47/76
• CPC Class Codes: H04L 65/1083 In-session procedures
• CPC Class Codes: H04L 65/1101 Session protocols
• CPC Class Codes: H04L 65/1104 Session initiation protocol [SIP]
• CPC Class Codes: H04L 65/70 Media network packetisation
• CPC Class Codes: H04L 65/80 Responding to QoS
• CPC Class Codes: H04M 7/0072 Speech codec negotiation H04L65/1069 takes precedence
• CPC Class Codes: H04W 88/181 Transcoding devices; Rate adaptation devices

Abstract

A system and method for dynamically adapting the transmission rate of packets in real-time voice over IP communications to the available bandwidth. A real-time bandwidth monitor (RTBM) for VoIP applications senses the call path bandwidth between two endpoints of a VoIP communication and adapts in real-time the packet transmission rate to utilize that bandwidth. If sufficient bandwidth is available, the RTBM selects a low compression, low latency CODEC to offer best possible voice quality to the user. If the bandwidth is constrained, the RTBM degrades gracefully by switching to a high compression CODEC. On further bandwidth reduction, the RTBM increases the media frames per packet. Because the bandwidth reduction may be transitory, the RTBM constantly monitors the end-to-end available bandwidth so as to invoke the CODEC/frame per packet combination that provides the best quality of service (QoS) achievable over the current end-to-end available bandwidth.

Claims

1. A method for adapting the transmission rate of media packets between endpoints in a voice over Internet protocol (VoIP) communication comprising:

determining a starting bandwidth measure at a starting endpoint;

selecting a staffing CODEC at the starting endpoint based on the starting bandwidth measure, wherein the starting CODEC is associated with a staffing CODEC nominal data rate;

determining an ending bandwidth measure at an ending endpoint;

selecting an ending CODEC at the ending endpoint based on the ending bandwidth measure, wherein the ending CODEC is associated with an ending CODEC nominal data rate;

informing the ending endpoint of the starting CODEC nominal data rate;

informing the starting endpoint of the ending CODEC nominal data rate;

selecting a current CODEC comprising a data rate equal to the lower of the starting CODEC nominal data rate and the ending CODEC nominal data rate;

using the current CODEC at the starting and ending end points;

obtaining a packet loss ratio of a media packet stream between the starting endpoint and the ending endpoint;

determining whether the packet loss ratio exceeds a maximum packet loss ratio associated with the current CODEC;

when the packet loss ratio exceeds the maximum packet loss ratio, determining a nominal in-use data rate of the current CODEC;

determining whether the current CODEC is associated with an alternate nominal data rate that is lower than the nominal in-use data rate;

when the current CODEC is associated with the alternate nominal data rate that is lower than the nominal in-use data rate, substituting the alternate nominal data rate for the nominal in-use data rate;

when the current CODEC is not associated with the alternate nominal data rate that is lower than the in-use nominal data rate, determining whether a current frames per packet measure is less than a maximum frames per packet measure associated with the current CODEC;

when the current frames per packet measure is less than the maximum frames per packet measure associated with the current CODEC, increasing the frames per packet measure of the media packet stream;

when the current frames per packet measure is greater than or equal to the maximum frames per packet measure, determining whether a substitute CODEC having a substitute nominal data rate that is lower than the in-use nominal data rate is available at the starting and ending endpoints; and

when the substitute CODEC is available at the staffing and ending endpoints, using the substitute CODEC at the staffing and ending endpoints.

2. The method for adapting the transmission rate of media packets between endpoints in a VoIP communication of claim 1, wherein determining the staffing bandwidth measure comprises:

sending a staffing probe packet from the staffing endpoint to a network device;

echoing the staffing probe packet from the network device to the staffing endpoint; and

determining the bandwidth of the path from the staffing endpoint to the network device.

3. The method for adapting the transmission rate of media packets between endpoints in a VoIP communication of claim 2, wherein the network device is selected from a STUN server, a SIP server, an endpoint, and an echo server.

4. The method for adapting the transmission rate of media packets between endpoints in a VoIP communication of claim 1, wherein selecting the starting CODEC at the staffing endpoint comprises:

associating the starting CODEC with a bandwidth range;

determining whether the staffing bandwidth measure is within the bandwidth range; and

if the staffing bandwidth measure is within the bandwidth range, then selecting the starting CODEC.