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

• US 20050201414A1
• Filed: 03/11/2005
• Published: 09/15/2005
• Est. Priority Date: 03/11/2004
• Status: Active Grant

• Current Assignee: Upland Software, Inc.
• Original Assignee: I2Telecom International Inc.
• Inventors: Awais, Ali
• Granted Patent: US 7,460,480 B2
• Time in Patent Office: Days
• US Class Current: 370/468
• 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 starting CODEC at the starting endpoint based on the starting bandwidth measure, wherein the starting CODEC is associated with a starting CODEC nominal data rate;

determining an ending bandwidth measure at the 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; and

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

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

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

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

determining the bandwidth of the path from the starting 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 starting endpoint comprises:

associating the starting CODEC with a bandwidth range;

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

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

5. The method for adapting the transmission rate of media packets between endpoints in a VoIP communication of claim 1, wherein the method further comprises:

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;

if the packet loss ratio exceeds the maximum packet loss ratio, then determining an 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;

if current CODEC is associated with an alternate nominal data rate that is lower than the in-use data rate, then substituting the alternate nominal data rate for the in-use nominal data rate.

6. The method for adapting the transmission rate of media packets between endpoints in a VoIP communication of claim 5, wherein the method further comprises:

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

if the current frames per packet measure is less than the maximum frames per packet measure associated with the current CODEC, then increasing the frames per packet measure of the media packet.

7. The method for adapting the transmission rate of media packets between endpoints in a VoIP communication of claim 6, wherein the method further comprises:

if the current frames per packet measure is greater than or equal to the maximum frames per packet, then 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

if the substitute CODEC is available at the starting and ending endpoints, then using the substitute CODEC at the starting and ending endpoints.