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- Aug 15, 2014
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We have been having horrendous audio quality issues and as such, we’ve been in troubleshooting mode. We're running V15 with the latest update, and our phones are almost entirely Yealink T46G (we have a few T46S now as well) We had some hardware issues at our Main office, and moved our phone system to our bigger VMWare Cluster at the Data Center. We are connected by several links (50 and 100 meg links) between our main office and where our Datacenter is. We are only at about 20-30% bandwidth use on any link. We record all the phone calls on virtually every extension. (We’ve disabled all but 10 recording extensions for troubleshooting). We have some new hardware here now, and we’re planning to bring the phone system back in house.
However, In the process of troubleshooting, we discovered a very , very disturbing issue. EVERY last drop of voice traffic is being funneled through the phone system. EVEN IF recording is disabled, even if there is no checkbox in the “PBX Delivers Audio” box ALL audio is transmitted via the 3CX Server. No boxes selected in troubleshooting at all and Everything still flows through the PBX.
90% of the phones are in the same building as our Patton PRI gateway. We have Patton FXS boxes for faxing, and all of that traffic is being funneled through the phone system as well. Only SIP should be going to the phone system, and all RTP traffic should be going between the phone and the PRI box. Even two extensions talking to each other (sitting next to each other on the same subnet in the same office) are going through the phone system.
We’ve captured this with Wireshark mirroring our entire voice vlan. Sure enough, every last conversation has to go down to the data center and back up. So it’s coming into the PRI Up here in our main office, then it goes down to the phone system, then back up to the extension up here in our main office. We have some SIP trunks that terminate into the phone system in the data center. We’d expect those to be funneled through the PBX since it’s all terminated there. Having all calls bounce up and down the WAN is highly suspect.
What is the reasoning for funneling all the voice traffic through the PBX eve if there is a SIP gateway and all our phones are talking SIP. Cisco, Altigen, Avaya work like they are supposed to and SIP is just for signaling. Endpoint to Endpoint talks RTP and bypasses the PBX for audio. Extension to Extension calls, not being recorded – there is no possible reason I can think of that with a VoIP phone system to have those calls go all the way back to the phone system and back out to the phones. It’s Crazy.
If anyone has an explanation, I would appreciate it. It’s counterintuitive and defeats the purpose of deploying the SIP/RTP combination.
Greg
However, In the process of troubleshooting, we discovered a very , very disturbing issue. EVERY last drop of voice traffic is being funneled through the phone system. EVEN IF recording is disabled, even if there is no checkbox in the “PBX Delivers Audio” box ALL audio is transmitted via the 3CX Server. No boxes selected in troubleshooting at all and Everything still flows through the PBX.
90% of the phones are in the same building as our Patton PRI gateway. We have Patton FXS boxes for faxing, and all of that traffic is being funneled through the phone system as well. Only SIP should be going to the phone system, and all RTP traffic should be going between the phone and the PRI box. Even two extensions talking to each other (sitting next to each other on the same subnet in the same office) are going through the phone system.
We’ve captured this with Wireshark mirroring our entire voice vlan. Sure enough, every last conversation has to go down to the data center and back up. So it’s coming into the PRI Up here in our main office, then it goes down to the phone system, then back up to the extension up here in our main office. We have some SIP trunks that terminate into the phone system in the data center. We’d expect those to be funneled through the PBX since it’s all terminated there. Having all calls bounce up and down the WAN is highly suspect.
What is the reasoning for funneling all the voice traffic through the PBX eve if there is a SIP gateway and all our phones are talking SIP. Cisco, Altigen, Avaya work like they are supposed to and SIP is just for signaling. Endpoint to Endpoint talks RTP and bypasses the PBX for audio. Extension to Extension calls, not being recorded – there is no possible reason I can think of that with a VoIP phone system to have those calls go all the way back to the phone system and back out to the phones. It’s Crazy.
If anyone has an explanation, I would appreciate it. It’s counterintuitive and defeats the purpose of deploying the SIP/RTP combination.
Greg