Device, system, and method of voice-over-IP communication

• US 8,831,001 B2
• Filed: 06/24/2012
• Issued: 09/09/2014
• Est. Priority Date: 06/24/2012
• Status: Active Grant

• Current Assignee: Audiocodes Limited
• Original Assignee: Audiocodes Limited
• Inventors: Klimker, Oren
• Primary Examiner(s): Yao, Kwang B
• Assistant Examiner(s): PATEL, HARDIKKUMAR D
• Application Number:
  US13/531,542
• Publication Number:
  US 20130343381A1
• Time in Patent Office: 807 Days
• Field of Search: None
• US Class Current: 370/392
• CPC Class Codes: H04L 47/10 Flow control; Congestion control
• CPC Class Codes: H04L 47/2416 Real-time traffic
• CPC Class Codes: H04L 47/28 in relation to timing considerations
• CPC Class Codes: H04L 47/34 ensuring sequence integrity, e.g. using sequence numbers
• 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 65/764 at the destination reformatting of additional data in video clients H04N21/4355
• CPC Class Codes: H04L 65/765 intermediate
• CPC Class Codes: H04L 65/80 Responding to QoS

Abstract

The present invention includes devices, systems, and methods of Voice-over-Internet Protocol (VoIP) communication. For example, a method includes: receiving a data stream comprising a set of VoIP packets; and modifying a Real Time Protocol (RTP) header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream. Optionally, the method includes decreasing the jitter buffer delay by: dropping at least one packet from said data stream; and decreasing a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet. Optionally, the method includes increasing the jitter buffer delay by: identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers; and increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets.

Claims

1. A method of Voice over Internet Protocol (VoIP) communication, the method comprising:

receiving a data stream comprising a set of VoIP packets; and

modifying a Real Time Protocol (RTP) header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream, wherein the method comprises increasing the jitter buffer delay of said data stream by performing;

(a) identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers;

(b) increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets, wherein the increasing comprises;

non-sequentially increasing the sequence number in the RTP header by a natural number computed based on a target increase in jitter buffer delay of a remote VoIP component to which a modified data stream is to be transmitted;

(c) transmitting to the remote VoIP component the modified data stream which comprises one or more packet gaps, to cause the remote VoIP receiver to recognize a packet loss and to remotely generate one or more additional VoIP packets using packet loss concealment.

2. The method of claim 1, comprising decreasing the jitter buffer delay by:

dropping at least one packet from said data stream; and

decreasing a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet.

3. The method of claim 2, comprising:

decreasing a sequence number and a timestamp value in RTP headers of all packets subsequent to said at least one packet.

4. The method of claim 1, comprising:

increasing a sequence number in RTP headers of all packets in said data stream starting at said latter packet and onward.

5. The method of claim 1, comprising:

modifying the RTP header without decoding a payload of any VoIP packet of said data stream.

6. The method of claim 1, comprising:

modifying the RTP header without re-encoding a payload of any VoIP packet of said data stream.

7. The method of claim 1, comprising:

detecting a portion of the data stream which corresponds to a silence period; and

modifying the RTP headers of one or more VoIP packets of that portion of the data stream more frequently relative to other one or more portions of said data stream.

8. An apparatus of Voice over Internet Protocol (VoIP) communication, the apparatus comprising:

a jitter buffer to receive a data stream comprising a set of VoIP packets; and

a Real Time Protocol (RTP) header modifier to modify an RTP header of at least one of said VoIP packets to modify a jitter buffer delay of said data stream,

wherein the RTP header modifier is to increase the jitter buffer delay by performing;

(a) identifying a pair of consecutive packets in the incoming data stream, the pair of consecutive packets having consecutive sequence numbers;

(b) increasing a sequence number in an RTP header of at least a latter packet in said pair of consecutive packets, wherein the increasing comprises;

non-sequentially increasing the sequence number in the RTP header by a natural number computed based on a target increase in jitter buffer delay of a remote VoIP component to which a modified data stream is to be transmitted;

(c) transmitting to the remote VoIP component the modified data stream which comprises one or more packet gaps, to cause the remote VoIP receiver to recognize a packet loss and to remotely one or more additional VoIP packets using packet loss concealment.

9. The apparatus of claim 8, wherein the RTP header modifier is to decrease the jitter buffer delay;

wherein the apparatus comprises a jitter buffer correction logic to drop at least one packet from said data stream;

wherein the RTP header modifier is to decrease a sequence number and a timestamp value in an RTP header of at least one additional packet subsequent to said at least one packet.

10. The apparatus of claim 9, wherein the RTP header modifier is to decrease a sequence number and a timestamp value in RTP headers of all packets subsequent to said at least one packet.

11. The apparatus of claim 8, wherein the RTP header modifier is to increase a sequence number in RTP headers of all packets in said data stream starting at said latter packet and onward.

12. The apparatus of claim 8, wherein the RTP header modifier is to modify the RTP header without decoding a payload of any VoIP packet of said data stream.

13. The apparatus of claim 8, wherein the RTP header modifier is to modify the RTP header without re-encoding a payload of any VoIP packet of said data stream.

14. The apparatus of claim 8, comprising a jitter buffer correction logic to detect a portion of the data stream which corresponds to a silence period;

and wherein the RTP header modifier is to modify the RTP headers of one or more VoIP packets of that portion of the data stream more frequently relative to other one or more portions of said data stream.

15. The apparatus of claim 8, wherein the jitter buffer is to receive a first data stream portion and a second, subsequent, data stream portion;

wherein for the first data stream portion, the RTP header modifier is to decrease the jitter buffer delay; and

wherein for the second data stream portion, the RTP header modifier is to increase the jitter buffer delay.

16. The apparatus of claim 8, wherein the apparatus comprises a session border controller.