Methods and systems for routing telecommunications
Methods and systems of routing calls over a network are described herein. A latency measurement is performed for at least a plurality of portions of a plurality of network paths and storing latency measurement information in computer readable memory. A dropped packet measurement is performed for at least portions of the plurality of network paths and storing dropped packet measurement information in computer readable memory. A call origination request is received. A network routing path is selected from the plurality of network paths, the network paths including a call manager, based at least in part on the latency measurement information and the dropped packet measurement information. A proxy system associated with a SIP provider is informed of the call. The call manager in the selected network path generates the call.
1. A method of routing SIP calls, the method comprising:
originating a first DTMF tone from a first call manager over a first network path, wherein the first call manager is not associated with a SIP network service provider that provides a first SIP network;
originating a second DTMF tone from a second call manager over a second network path, wherein the second call manager is not associated with the SIP network service provider;
performing a latency measurement for at least a portion of the first network path by comparing a time delay from the origination of the first DTMF tone to receipt of the first DTMF tone by the first call manager and storing corresponding first network path latency measurement information in computer readable memory;
performing a dropped packet measurement for at least the portion of the first network path based on discontinuities or gaps in the first DTMF tone when received back by the first call manager and storing corresponding first network path dropped packet measurement information in computer readable memory;
performing a latency measurement for at least a portion of the second network path, by comparing a time delay from the origination of the second DTMF tone to receipt of the second DTMF tone by the second call manager and storing corresponding second network path latency measurement information in computer readable memory;
performing a dropped packet measurement for at least the portion of the second network path, the path based on discontinuities or gaps in the second DTMF tone when received back by the second call manager and storing corresponding second network path dropped packet measurement information in computer readable memory;
receiving an inbound call at the first SIP network, wherein the inbound call is directed to a phone number assigned to an operator not associated with the SIP network provider;
calculating a network quality score for each of the first and second call managers using a formula including:
a first weighting related to latency;
the first network path latency measurement information for the first network path,
wherein the first network path latency information is adjusted using the first weighting when calculating the network quality score for the first call manager;
the second network path latency measurement information,
wherein the second network path latency measurement information for the second network path is adjusted using the first weighting when calculating the network quality score for the second call manager;
a second weighting related to dropped packets, where the first weighting is different than the second weighting;
the first network path dropped packet measurement information,
wherein the first network path dropped packet measurement information is adjusted using the second weighting when calculating the network quality score for the first call manager;
the second network path dropped packet measurement information,
wherein the second network path dropped packet measurement information is adjusted using the second weighting when calculating the network quality score for the second call manager;
selecting, using a computing device, the first call manager or the second call manager to process the inbound call based at least in part on the network quality score for the first call manager and the network quality score for the second call manager;
terminating the inbound call at the selected call manager,
wherein the selected call manager is configured to perform one or more of call answering, call screening, call forwarding, call bridging, or call conferencing applications; and
receiving at the selected call manager packetized voice media associated with the inbound call from a media gateway associated with the SIP network service provider.
2. The method as defined in claim 1 , wherein the selected call manager is selected based in part on a media content type.
3. The method as defined in claim 1 , wherein the selected call manager is selected based in part on a priority of a caller or a called party associated with the inbound call.
4. The method as defined in claim 1 , wherein the selected call manager is selected based in part on available network bandwidth from the selected call manager to the media gateway.
5. The method as defined in claim 1 , wherein a first SIP proxy performs the call manager selection.
6. The method as defined in claim 1 , wherein the inbound call is directed to a phone number associated with an operator of the selected call manager.
7. The method as defined in claim 1 , the method further comprising selecting the first call manager or the second call manager to process the inbound call based in part on measured jitter in a network path between the first call manager and the media gateway.
8. The method as defined in claim 1 , wherein the selected call manager is selected based in part on whether media presented by the inbound call is voice media or video media.
9. The method as defined in claim 1 , wherein the selected call manager is selected based in part on a relative priority of the inbound call as determined by the phone number to which the inbound call is directed to.