RTP Media

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biggreyphone

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I am using a 3CX hosted system with local 3CX SBC. I am using flowroute for my SIP trunks and had a few calls where there was no audio. I had Flowroute trace the calls and they said it appeared there were no issues. Flowroute indicates that they use the method listed below for direct audio delivery. In using an SBC, do I need a firewall rule for my router or does use of an SBC eliminate the need for this?

RTP media (call audio): To reduce latency, Flowroute uses Direct Media Delivery. To receive Direct Audio, allow UDP packets from any source IP address with a destination port within your system's RTP public media port range
 
You do not need any firewall inbound rules as you using a sbc.

Just make sure you allow sbc server outbound traffic - 3cx tunnel (udp and tcp port 5090) and https port (443)

What phones are you using ?
 
You do not need any firewall inbound rules as you using a sbc.

Just make sure you allow sbc server outbound traffic - 3cx tunnel (udp and tcp port 5090) and https port (443)

What phones are you using ?
Fanvil.
 
There was a new template released in the last 24hrs, you could try this - check updates in console
 
I am using a 3CX hosted system with local 3CX SBC. I am using flowroute for my SIP trunks and had a few calls where there was no audio. I had Flowroute trace the calls and they said it appeared there were no issues. Flowroute indicates that they use the method listed below for direct audio delivery. In using an SBC, do I need a firewall rule for my router or does use of an SBC eliminate the need for this?

RTP media (call audio): To reduce latency, Flowroute uses Direct Media Delivery. To receive Direct Audio, allow UDP packets from any source IP address with a destination port within your system's RTP public media port range
SBC or no SBC, the RTP from Flowroute doesn't go to the extension, it goes to the PBX. If you look at the trunk settings, you will see 'PBX Delivers Audio' is on meaning the PBX stays in the call path. As far as what Flow Route is saying is make sure you don't restrict the RTP ports to only the SIP addresses as they will get out of the call path and let the audio go directly from the PSTN to your 3CX instance and not route through Flowroute once the call is established. Since you are on 3CX hosted and can't change the firewall rules, this isn't a problem.
 
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I am using a 3CX hosted system with local 3CX SBC. I am using flowroute for my SIP trunks and had a few calls where there was no audio. I had Flowroute trace the calls and they said it appeared there were no issues. Flowroute indicates that they use the method listed below for direct audio delivery. In using an SBC, do I need a firewall rule for my router or does use of an SBC eliminate the need for this?

RTP media (call audio): To reduce latency, Flowroute uses Direct Media Delivery. To receive Direct Audio, allow UDP packets from any source IP address with a destination port within your system's RTP public media port range
You don't need to switch phones or change your firewall! You simply need to open the necessary RTP ports from Flowroute to your 3CX server. That's all. Send me a DM if you're still struggling with this.
 
Just want to clarify that, as cobaltit already said, if the instance is Hosted by 3CX, you cannot edit the firewall rules but then again there is no need as all port forwarding should be already taken care of.

If this is something that keeps happening I recommend trying to narrow it down a bit more with questions like, is this only affecting external calls or internal too, happening for only one remote site, is audio lost both ways, etc. That way, if there's a problem, we'll at least know where to start looking.
 
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