Voip, dropped calls.
Client of ours has had problems with VOIP problems for a good long time. We've done quite a bit to try to fix the problems, but we still have dropped calls, and haven't found a reason for them.
They're using a Hosted PBX running 3cx, which is not behind a NAT.
They're also using a dedicated onsite SBC (3cxSBC), which only runs the SBC application. (Windows 7 PRO x64, plenty of resources).
We had them running on a Sonicwall previously, but after enough recommendations, segregated their entire phone system onto it's own switch/router (netgear FS728, and Cisco RV series router.)
They're using sotel systems for their SIP trunking service, Yealink T46 and 41G phones, and regularly encounter issues like the phones dropping calls, or being unable to place outbound calls. Their Desktop HUDs will typically show "problem connecting to 3cx tunnel".
We've opened tickets with several support vendors, including the PBX software vendor, the hosted server vendor, and have outsourced a little as well with VOIP specialists.
In one recent example, a call was dropped after 3 minutes in duration, and audio was live on both sides of the call at the time of the drop-out. The sip trunk provider states that the call (outbound) was dropped on the recipient side, by the carrier. SIP-ALG is not enabled on the router itself, and there is no need for port forwarding/etc. If anyone has any interest in this, please let me know and I can give you any more information that you might need, packet captures ,etc.
They're using a Hosted PBX running 3cx, which is not behind a NAT.
They're also using a dedicated onsite SBC (3cxSBC), which only runs the SBC application. (Windows 7 PRO x64, plenty of resources).
We had them running on a Sonicwall previously, but after enough recommendations, segregated their entire phone system onto it's own switch/router (netgear FS728, and Cisco RV series router.)
They're using sotel systems for their SIP trunking service, Yealink T46 and 41G phones, and regularly encounter issues like the phones dropping calls, or being unable to place outbound calls. Their Desktop HUDs will typically show "problem connecting to 3cx tunnel".
We've opened tickets with several support vendors, including the PBX software vendor, the hosted server vendor, and have outsourced a little as well with VOIP specialists.
In one recent example, a call was dropped after 3 minutes in duration, and audio was live on both sides of the call at the time of the drop-out. The sip trunk provider states that the call (outbound) was dropped on the recipient side, by the carrier. SIP-ALG is not enabled on the router itself, and there is no need for port forwarding/etc. If anyone has any interest in this, please let me know and I can give you any more information that you might need, packet captures ,etc.
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How much bandwidth?
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It's a 50/10 pipe from Comcast Business.
It's 17 phones, no more than 4 phone calls are ever taking place at one time.
I can't imagine a Fairly Decently powered Sonicwall or newer Cisco RV320 causing the problems, though multiple people have stated they have. The majority of calls sound great on both ends of the line, and the load on the device and circuit are both minor.
The circuit is being shared by their data/voice network, but there is bandwidth limitations on the data network side which should leave plenty of headroom for the calls themselves.
I've just recently setup emco ping monitor to get constant pings out to a few hosts to see if their dropped calls are synced with spikes or packet loss. I'm not looking for a UTM to replace the device, but the Microtik's are so low priced that I'd consider one I suppose. Really surprised though about the RV320, the vendor spoke about the RV routers as though they would be perfect for this particular job.
It's 17 phones, no more than 4 phone calls are ever taking place at one time.
I can't imagine a Fairly Decently powered Sonicwall or newer Cisco RV320 causing the problems, though multiple people have stated they have. The majority of calls sound great on both ends of the line, and the load on the device and circuit are both minor.
The circuit is being shared by their data/voice network, but there is bandwidth limitations on the data network side which should leave plenty of headroom for the calls themselves.
I've just recently setup emco ping monitor to get constant pings out to a few hosts to see if their dropped calls are synced with spikes or packet loss. I'm not looking for a UTM to replace the device, but the Microtik's are so low priced that I'd consider one I suppose. Really surprised though about the RV320, the vendor spoke about the RV routers as though they would be perfect for this particular job.
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ASKER
Bryant, doing exactly that, have no problem with running emco in the background for a while to see if the issues sync up with circuit issues. Haven't been pinging the sip gateway of the provider, maybe I can get that information from them and see.
Cliff, I like ubiquiti and have sold some of their stuff, not familiar enough with vyatta to feel super comfortable with the edgerouters though their specs are great for the price.
Haven't used microtik personally but have heard enough good about them. The pick up of the RV320 was a recommendation, though perhaps not a great one. I'll see about picking up a microtik and testing it.
Cliff, I like ubiquiti and have sold some of their stuff, not familiar enough with vyatta to feel super comfortable with the edgerouters though their specs are great for the price.
Haven't used microtik personally but have heard enough good about them. The pick up of the RV320 was a recommendation, though perhaps not a great one. I'll see about picking up a microtik and testing it.
ASKER
Another thought has been that perhaps the Modem itself is causing problems. It's Comcast, the modem is rented, and they have a block of 5 static IP addresses. I don't know of any modems that are recommended for VOIP that we'll be able to use with comcast, as they won't be able to configure the static ip addresses.
Packet captures both coming into their network, and hitting the pbx itself, there's very constant TCP resets (there are not on the other clients using the same PBX) as well as malformed SIP headers.
First guess of course is SIP-ALG, we've tried several routers to no avail, thoughts for the tcp reset's were network congestion which we can very much rule out as we've tried different switches, etc.
The modem is a variable which presents a challenge- what we can use, with comcast, with a block of static ip addresses, seems to be fairly limited.
Any thoughts on this would be hugely appreciated. Can share packet captures if anyone is interested.
Packet captures both coming into their network, and hitting the pbx itself, there's very constant TCP resets (there are not on the other clients using the same PBX) as well as malformed SIP headers.
First guess of course is SIP-ALG, we've tried several routers to no avail, thoughts for the tcp reset's were network congestion which we can very much rule out as we've tried different switches, etc.
The modem is a variable which presents a challenge- what we can use, with comcast, with a block of static ip addresses, seems to be fairly limited.
Any thoughts on this would be hugely appreciated. Can share packet captures if anyone is interested.
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Question answered
ASKER
What's a model Microtik that you recommend for a 20 remote phone network