Understanding RTP Sound Issues

Hi Guys,

I am using Freepbx, most oft the time it is working nice and smooth.
Sometimes i receive a call and after a few seconds we have only 1-way Audio. i can’t hear the external caller anymore.
Sometimes it happens after a few seconds. Sometimes after 2mins …
Most of the time it is working without issues.

i found this https://www.google.de/search?q=one+way+audio+fix+freepbx&oq=one+way+audio+fix+freepbx&aqs=chrome..69i57.4980j0j7&sourceid=chrome&ie=UTF-8

we are using as Router FritzBox 7580.
I have forwared port 5060 -> to my PBX.

Is it needed to forward also the RTP Ports to my pbx ?

I have a DECT Station and SNOM Phones.
The Phone and DectStation also are having setup for RTP Port Ranges.
Do i have to use the same ranges in my snom like in my PBX or is it better to use difrrent ranges?
Is the Portforward only needed to my PBX or also to my SNOM Phone / Dect station ?

Hope someone can make this clear.

assuming you have sip trunks and/or remote extensions then yes you should forward UDP 10000-20000 (freepbx default RTP range)

you can/should lock down UDP 5060 to the IP(s) of your trunk provider but unless they proxy media, the RTP range should be left open; in the case of a remote extension you can lock both down the the remote stations public gateway address

Hello Dolesec, thank you for your reply.

yes i have SIP trunks. The Extensions are all within my LANsubnet (same LAN like freepbx!).

[quote=“dolesec, post:2, topic:44656”]
the RTP range should be left open;
[/quote] you mean i should open and forward the RTP Media Ports (10000-20000) to my FreePBX right ?
and also port 5060 to my Freepbx ? right ?

What about DYNAMIC RTP Ports of my Snom Phones, are these important relating to FreePBX

Should I use the same Ports like in my FreePBX or doesn’t this matter?
What about if i have multiple Snom Phones ? Can i Use the same dnymaic RTP Ports or should i Use diffrent then in my FreePBX.

It would be nice to get a clear explanation of this.

leave the phones alone - they only speak with the pbx and and will negotiate that with the pbx

the pbx however will use the rtp range speicified in its dialogue with the trunking servers and thus yes you need to forward that range to the PBX

this should be crystal clear at this point ?
if not let me know

  1. Leave the phones alone as dolesec said.

  2. You do not open or forward any RTP ports from your provider or the remote side. That is completely unnecessary and un-needed.

  3. FritzBoxes have built in VoIP systems with media services like voicemail, IVR, etc. You really need to make sure that it is not causing an issue with your traffic. You basically have another “PBX” between you and the outside.It could be sending re-INVITEs or answering re-INVITES when it shouldn’t and taking over the RTP stream.

The RTP ports are not used during the SDP offer/answer of the call setup. It’s not until after a 18X reply will RTP be used (if early media) but not until after the 200 OK reply for sure. Having NAT rules for RTP is for your side of the equation to make sure your router doesn’t close up NAT holes. Overall if you have a decent enough router, you don’t need them.

You never, ever put in NAT rules for the ITSP/Carriers RTP. That’s just silly.

sorry if my statement was confusing - thats why i asked if it was clear

I beleive Tom misunderstood

anyway, i have no idea what a fritzbox is - my statements apply if you’re dealing with a typical routing setup without any special handling of sip and using external sip trunks

nothing i said about RTP was targeted at the provider specifc port settings so not sure how that was interpreted … my point was to limit spurious traffic one can lock down 5060 to the providers trunk IP’s; however you cannot restrict RTP ports UNLESS that same provider is prozying media via those same IP’s - so not sure how you interpreted what i said befiore but i hope this makes it more clear

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