Core network interoperability in a pico cell system
AssigneeQUALCOMM Incorporated
InventorsSergio Kolor, Andrew Hunter, Michael Green, Nikhil Jain, Qiuzhen Zou
Life of this patent
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Abstract
In a wireless communication system, a method and apparatus of communicating between at least one base station and a core network is claimed. Signaling messages and voice streams are transported using packet-switched protocols (IP). A CIC field (Circuit Identity Code) is used to identify at least one VoIP streams. The CIC field may be interpreted as a UDP port number for the voice over IP stream. Also, a synchronization source or contributing source filed of a RTP header may be used as the CIC field.
Claims24
A method for transporting a first and a second voice-over-internet-protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-Time Transport Protocol (RTP) protocol, comprising the steps of:
- pre-assigning a same User Datagram Protocol (UDP) port number to the Base Station Controller and the Media Gateway;
- selecting a first Circuit Identity Code for a first VoIP stream;
- selecting a second Circuit Identity Code for a second VoIP stream;
- inserting the first Circuit Identity Code into a field of a first RTP header;
- inserting the second Circuit Identity Code into a field of a second RTP header;
- multiplexing the first and second VOIP streams together;
- transmitting the first and second VoIP streams to the pre-assigned UDP port number at the Base Station Controller;
- wherein inserting the first Circuit Identify Code into the field of the first RTP header is for a first data packet of the first VoIP stream, and wherein inserting the second Circuit Identity Code into the field of the second RTP header is for a second data packet of the second VoIP stream;
- and wherein the field of the first RTP header and the field of the second RTP header comprise a Synchronization Source field having a 32-bit length, wherein inserting the first Circuit Identity Code into the field of the first RTP header and wherein inserting the second Circuit Identity Code into the field of the second RTP header both further comprise generating the Synchronization Source field using 16 lower significant bits of the 32-bit length field.
A computer program product, stored in a memory, for transporting a first and a second voice-over-internet protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-Time Transport Protocol (RTP) protocol, comprising:
- at least one instruction for pre-assigning a same User Datagram Protocol (UDP) port number to the Base Station Controller and the Media Gateway;
- at least one instruction for selecting a first Circuit Identity Code for a first VoIP stream;
- at least one instruction for selecting a second Circuit Identity Code for a second VoIP stream;
- at least one instruction for inserting the first Circuit Identity Code into a field of a first RTP header;
- at least one instruction for inserting the second Circuit Identity Code into a field of a second RTP header;
- at least one instruction for multiplexing the first and second VoIP streams together;
- at least one instruction for transmitting the first and second VoIP Streams to the pre-assigned UDP port number at the Base Station Controller;
- wherein the at least one instruction for inserting the first Circuit Identity Code into the field of the first RTP header is for a first data packet of the first VoIP stream, and wherein the at least one instruction for inserting the second Circuit Identity Code into the field of the second RTP header is for a second data packet of the second VoIP stream;
- and wherein the field of the first RTP header and the field of the second RTP header comprise a Synchronization Source field having a 32-bit length, wherein inserting the first Circuit Identity Code into the field of the first RTP header and wherein inserting the second Circuit Identity Code into the field of the second RTP header both further comprise generating the Synchronization Source field using 16 lower significant bits of the 32-bit length field.
An apparatus for transporting a first and a second voice-over-internet protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-lime Transport Protocol (RTP) protocol, comprising:
- means for pre-assigning a same User Datagram Protocol (UDP) port number to the Base Station Controller and the Media Gateway;
- means for selecting a first Circuit Identity Code for a first VoIP stream;
- means for selecting a second Circuit Identity Code for a second VoIP stream;
- means for inserting the first Circuit Identity Code into a field of a first RTP header;
- means for inserting the second Circuit Identity Code into a field of a second RTP header;
- means for multiplexing the first and second VOIP streams together;
- means for transmitting the first and second VoIP streams to the pre-assigned UDP port number at the Base Station Controller;
- wherein inserting the first Circuit Identify Code into the field of the first RTP header is for a first data packet of the first VoIP stream, and wherein inserting the second Circuit Identity Code into the field of the second RTP header is for a second data packet of the second VoIP stream;
- and wherein the field of the first RTP header and the field of the second RTP header comprise a Synchronization Source field having a 32-bit length, wherein inserting the first Circuit Identity Code into the field of the first RTP header and wherein inserting the second Circuit Identity Code into the field of the second RTP header both further comprise generating the Synchronization Source field using 16 lower significant bits of the 32-bit length field.
A method for transporting a first and a second voice-over-internet-protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-Time Transport Protocol (RTP) protocol, comprising the steps of:
- receiving RTP data packets at a pre-defined User Datagram Protocol (UDP) port number, the RTP data packets comprising a first VoIP stream, and a second VoIP stream;
- identifying the first VoIP stream if a field of an RTP header comprises a first Circuitry Identity Code;
- identifying the second VoIP stream if the field comprises a second Circuitry Identity Code;
- constructing the first VoIP stream from data within an RTP data packet having the first Circuitry Identity Code;
- constructing the second VoIP stream from data within an RTP data packet having the second Circuitry identity Code;
- transmitting the first and second VoIP streams to a same pre-defined UDP port number at the Base Station Controller;
- and
- wherein the field comprises a Synchronization Source field having a 32-bit length with the first Circuit Identity Code or the second Circuit Identity Code using 16 lower significant bits of the 32-bit length field.
A computer program product, stored in a memory, for transporting a first and a second voice-over-internet protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-Time Transport Protocol (RTP) protocol, comprising:
- at least one instruction for receiving RTP data packets at a pre-defined User Datagram Protocol (UDP) port number, the RTP data packets comprising a first VoIP stream, and a second VoIP stream;
- at least one instruction for identifying the first VoIP stream if a field of an RTP header comprises a first Circuitry Identity Code;
- at least one instruction for identifying the second VoIP stream if the field comprises a second Circuitry Identity Code;
- at least one instruction for constructing the first VoIP stream from data within an RTP data packet having the first Circuitry Identity Code;
- at least on instruction for constructing the second VoIP stream from data within an RTP data packet having the second Circuitry Identity Code;
- at least one instruction for transmitting the first and second VoIP streams to a same pre-defined UDP port number at the Base Station Controller;
- and wherein the field comprises a Synchronization Source field having a 32-bit length with the first Circuit Identity Code or the second Circuit Identity Code using 16 lower significant bits of the 32-bit length field.
An apparatus for transporting a first and a second voice-over-internet-protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-Time Transport Protocol (RTP) protocol, comprising:
- means for receiving RTP data packets at a pre-defined User Datagram Protocol (UDP) port number, the RTP data packets comprising a first VoIP stream and a second VoIP stream;
- means for identifying the first VoIP stream if a field of an RTP header comprises a first Circuitry Identity Code and for identifying the second VoIP stream if the field comprises a second Circuitry Identity Code;
- means for constructing the first VoIP stream from data within an RTP data packet having the first Circuitry Identity Code and for constructing the second VoIP stream from data within an RTP data packet having the second Circuitry Identity Code;
- means for transmitting the first and second VoIP streams to a same pre-defined UDP port number at the Base Station Controller;
- and
- wherein the field comprises a Synchronization Source field having a 32-bit length with the first Circuit Identity Code or the second Circuit Identity Code using 16 lower significant bits of the 32-bit length field.
At least one processor configured to transport a first and a second voice-over-internet protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-Time Transport Protocol (RTP) protocol, executing the instructions of:
- pre-assigning a same User Datagram Protocol (UDP) port number to the Base Station Controller and the Media Gateway;
- selecting a first Circuit Identity Code for a first VoIP stream;
- selecting a second Circuit Identity Code for a second VoIP stream;
- inserting the first Circuit Identity Code into a field of a first RTP header;
- inserting the second Circuit Identity Code into a field of a second RTP header;
- multiplexing the first and second VOIP streams together;
- transmitting the first and second VoIP streams to the pre-assigned UDP port number at the Base Station Controller;
- wherein inserting the first Circuit Identify Code into the field of the first RTP header is for a first data packet of the first VoIP stream, and wherein inserting the second Circuit Identity Code into the field of the second RTP header is for a second data packet of the second VoIP stream;
- and
- wherein the field of the first RTP header and the field of the second RTP header comprise a Synchronization Source field having a 32-bit length, wherein inserting the first Circuit Identity Code into the field of the first RTP header and wherein inserting the second Circuit Identity Code into the field of the second RTP header both further comprise generating the Synchronization Source field using 16 lower significant bits of the 32-bit length field.
At least one processor configured to transport a first and a second voice-over-internet protocol (VoIP) stream between a Base Station Controller and a Media Gateway using at least a Real-Time Transport Protocol (RTP) protocol, executing the instructions of:
- receiving RTP data packets at a pre-defined User Datagram Protocol (UDP) port number, the RTP data packets comprising a first VoIP stream, and a second VoIP stream;
- identifying the first VoIP stream if a field of an RTP header comprises a first Circuitry Identity Code;
- identifying the second VoIP stream if the field comprises a second Circuitry Identity Code;
- constructing the first VoIP stream from data within an RTP data packet having the first Circuitry Identity Code;
- constructing the second VoIP stream from data within an RTP data packet having the second Circuitry Identity Code;
- transmitting the first and second VoIP streams to a same pre-defined UDP port number at the Base Station Controller;
- and wherein the field comprises a Synchronization Source field having a 32-bit length with the first Circuit Identity Code or the second Circuit Identity Code using 16 lower significant bits of the 32-bit length field.
Dependencies are inferred from the claim text.