Cross-Layer Optimization of VoIP Services in Advanced Wireless Networks

• US 20090109893A1
• Filed: 10/31/2007
• Published: 04/30/2009
• Est. Priority Date: 10/31/2007
• Status: Active Grant

• Current Assignee: Futurewei Technologies Incorporated (Huawei Investment & Holding Co., Ltd.)
• Original Assignee: Futurewei Technologies Incorporated (Huawei Investment & Holding Co., Ltd.)
• Inventors: Gopal, Thawatt
• Granted Patent: US 7,940,655 B2
• Time in Patent Office: Days
• US Class Current: 370/315
• CPC Class Codes: H04L 65/1016 IP multimedia subsystem [IMS]
• CPC Class Codes: H04L 65/1104 Session initiation protocol [SIP]
• CPC Class Codes: H04L 65/80 Responding to QoS

Abstract

A cross-layer optimization is presented between the MAC and Network layers to enhance the performance of VoIP services in advanced wireless networks. Among the factors considered in the optimization are the characteristics of the originating source and the network path. This information is relayed using the Network layer to provide feedback to the MAC layer in order to improve the end-to-end performance of VoIP services.

Claims

1. A method for managing voice over internet protocol (VoIP) services comprising:

deriving one or more characteristics of a source voice transmission from a first communication device in an originating network;

formatting said source voice transmission into VoIP transmission data;

adjusting one or more VoIP transmission parameters based on said one or more characteristics;

encoding said one or more characteristics into said VoIP transmission data;

transmitting said encoded VoIP transmission data to a terminating network;

decoding said one or more characteristics from said encoded VoIP transmission data; and

scheduling delivery of said source voice transmission to a second communication device based in part on said one or more characteristics, wherein at least one of said first and second communication devices is a wireless communication device.

2. The method of claim 1 wherein said one or more characteristics comprise one or more of:

a delay at said originating network;

a time of origin of said source voice transmission;

an estimated time of origin of said source voice transmission;

or

a type of voice transmission.

3. The method of claim 2 further comprising:

using one of said delay, said time of origin, or said estimated time of origin to calculate a delay time attributable to said originating network.

4. The method of claim 2 wherein said time of origin or said estimated time of origin are measured using a global synchronized clock, said method further comprising:

measuring a total absolute delay of said source voice transmission from said time of origin or estimated time of origin using said global synchronized clock, wherein said scheduling uses said total absolute delay.

5. The method of claim 2 wherein said type of voice transmission comprises one of:

a mobile to mobile call;

a land to mobile call;

or

a mobile to land call.

6. The method of claim 5, wherein said type of voice transmission is said mobile to land call;

wherein said adjusting occurs at said originating network and comprises;

updating one or more reverse link media access control (MAC) parameters to increase a delay interval for transmissions over an air interface between said first communication device and a base transceiver.

7. The method of claim 6 wherein said updating is determined based on one of:

a destination IP address corresponding to said terminating network comprising a public switched telephone network (PSTN);

or

a universal resource indicator (URI) field in a session initiation protocol (SIP) invite message corresponding to a PSTN communication device belonging to said terminating network comprising said PSTN.

8. The method of claim 6 wherein said updating comprises one or both of:

adjusting one or more hybrid automatic retransmission request (H-ARQ) terminating targets;

or

adjusting bandwidth allocation of said originating network.

9. The method of claim 6, wherein said type of voice transmission is said land to mobile call, said method further comprising:

receiving a switched-based transmission at a multimedia gateway from said first communication device of said originating network;

converting said switched-based transmission into said VoIP transmission data at said multimedia gateway;

wherein said deriving, said adjusting, said encoding, said decoding, and said scheduling occur at said terminating network.

10. The method of claim 1 further comprising:

calculating a delay time attributable to said originating network using said one or more characteristics; and

comparing said delay time with a maximum delay allowed by a host communication system, wherein said scheduling is conducted in accordance with said comparing.

11. The method of claim 1 wherein said first and second communication devices comprise said wireless communication device, and wherein said encoding comprises:

encoding said one or more characteristics into a fragment offset field of an internet protocol, version 4 (IPv4) header.

12. The method of claim 1 wherein said first and second communication devices comprise said wireless communication device, and wherein said encoding comprises:

encoding a realtime protocol (RTP) sequence number of said VoIP transmission data and said one or more characteristics into a realtime control protocol application (APP RTCP) packet.

14. The method of claim 1 wherein said deriving comprises:

examining the physical (PHY) and media access control (MAC) layers of said VoIP packet to determine said one or more characteristics.

13. The method of claim 13 wherein said one or more characteristics comprise said time of origin or said estimated time of origin determined using a global synchronized clock.

15. A communication system comprising:

a first network node, said first originating network node comprising;

a first processor;

a detection component operable by said processor that examines one or more of a physical (PHY) layer or a media access control (MAC) layer of an IP packet for transmission characteristics, said IP packet received from an originating communication device; and

a first interface that transmits said IP packet and said transmission characteristics to a second network node; and

said second network node comprising;

a second processor;

an encoder operable by said processor that encodes said transmission characteristics into said IP packet; and

an output interface that transmits the encoded IP packet onto a transmission network addressed to a terminating communication device, wherein at least one of said originating or terminating communication devices is a mobile communication device.

16. The communication system of claim 15 wherein said transmission characteristics comprise one of:

a delay at said originating network;

a time of origin of said source IP transmission;

an estimated time of origin of said source IP transmission;

or

a type of IP voice transmission

17. The communication system of claim 15 further comprising:

a third network node comprising;

a third processor;

an input interface that receives said encoded IP packet; and

a decoder operable by said third processor that extracts the transmission characteristics from said encoded IP packet; and

a fourth network node comprising;

a fourth processor; and

a scheduler that uses the transmission characteristics to schedule delivery of said IP packet to said terminating communication device.

18. A computer program product having a computer readable medium with computer program logic recorded thereon, said computer program product comprising:

code for deriving one or more characteristics of a source voice transmission from a first communication device in an originating network;

code for formatting said source voice transmission into VoIP transmission data;

code for adjusting one or more VoIP transmission parameters based on said one or more characteristics;

code for encoding said one or more characteristics into said VoIP transmission data;

code for transmitting said encoded VoIP transmission data to a terminating network;

code for decoding said one or more characteristics from said encoded VoIP transmission data; and

code for scheduling delivery of said source voice transmission to a second communication device based in part on said one or more characteristics, wherein at least one of said first and second communication devices is a wireless communication device.

19. The computer program product of claim 18 wherein said one or more characteristics comprise one or more of:

a delay at said originating network;

a time of origin of said source voice transmission;

an estimated time of origin of said source voice transmission;

or

a type of voice transmission.

20. The computer program product of claim 19 wherein said time of origin or said estimated time of origin are measured using a global synchronized clock, said method further comprising:

code for measuring a total absolute delay of said source voice transmission from said time of origin or estimated time of origin using said global synchronized clock, wherein said code for scheduling uses said total absolute delay.

21. The computer program product of claim 19 wherein said type of voice transmission comprises one of:

a mobile to mobile call;

a land to mobile call;

or

a mobile to land call.

22. The computer program product of claim 21, wherein said type of voice transmission is said mobile to land call;

wherein said adjusting occurs at said originating network and comprises;

code for updating one or more reverse link media access control (MAC) parameters to increase a delay interval for transmissions over an air interface between said first communication device and a base transceiver.

23. The computer program product of claim 22 wherein said code for updating is determined based on one of:

a destination IP address corresponding to said terminating network comprising a public switched telephone network (PSTN);

or

a universal resource indicator (URI) field in a session initiation protocol (SIP) invite message corresponding to said terminating network comprising said PSTN.

24. The computer program product of claim 22, wherein said code for updating comprises one or both of:

code for adjusting one or more hybrid automatic retransmission request (H-ARQ) terminating targets;

or

code for adjusting bandwidth allocation of said originating network.

25. The computer program product of claim 21, wherein said type of voice transmission is said land to mobile call and wherein said code for deriving, said code for adjusting, said code for encoding, said code for decoding, and said code for scheduling are executed at said terminating network.