Patent ID: 7852749
Filing Date: 2010-12-14
Classification: H04L,H04M

Abstract:
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: 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, 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.