20-01-2017 09:31 AM
I'm struggling here to get incoming voip audio to work. I dial a number on the IP phone and it connects and the line rings. When the other end picks up, I can't hear them. But if I call the voip phone from my mobile it works fine.
Thanks to anyone whwo can fix it in advance.
20-01-2017 01:17 PM
20-01-2017 01:35 PM
It depends a bit on the protocol. The commonest protocol is SIP over UDP. The control packets are usually sent and received on port 5060/udp. You then also need a bunch of RTP audio stream addresses, usually two per concurrent call supported. These can be anywhere (eg 10000-10100) and need to be forwarded to your VoIP phone otherwise you get one-way voice issues. The RTP port range has to be the same as configured on the VoIP phone. Which make of VoIP phone is it? I have a Gigaset S850 with a number of DECT phones and it works fine.
The SIP control channel can also be over TCP or encrypted over TLS but if the issue is one-way-voice it's most likely the RTP streams.
20-01-2017 08:04 PM
Thanks for responding. I am using a SIP service provided by localphone.com.
Port 5060 is forwarded to the ip phone, 192.168.1.189
A simple call between between the IP phone and the BT landline works fine. But when I call a company (most calls) I can never hear them. As soon as the call is answered, nothing but silence.
The IP phone is a Grandstream GXP 1620.
How do I configure the RTP port range? Given that they could be anything??
21-01-2017 07:59 AM - edited 21-01-2017 08:27 AM
Edit: read the two newer posts above first. You should be able to confirm the port from the Web page and then forward it on the Brightbox.
I read the user manual and they don't seem to be clear about the RTP settings. You may need to read up on STUN servers and enable one. They help it negotiate NAT. Also if your router has a setting for SIP ALG then change it (turn on if off and vice versa). Failing that I would end up resorting to tcpdump or Wireshark to see what ports it's using for RTP.
21-01-2017 08:11 AM
Try forwarding port 5004 UDP as well as 5060 UDP to the phone. Grandstream mention this in places as the RTP port.
21-01-2017 08:24 AM
There is an article on configuring RTP here: https://support.gradwell.com/hc/en-gb/articles/215556023-Setting-up-port-forwarding-on-a-Grandstream...
26-01-2017 10:19 AM
A quick update....all seems to be working properly now. Here is a summary of the setup for others:
(Note EE brightbox 2 does not have an ALG setting that you can change)
Router forwards ports 5004 TCP&UDP, 5060 - 5061 TCP&UDP to the phone
Ports 16600 - 16998 UDP are also forwarded for RTP but I think this might be unecessary as the RTP range is wider than that.
In the phone the following setup was added:
Check SIP User ID for Incoming INVITE = YES
Accept Incoming SIP from Proxy Only = YES
by JAMESDNEVILLE yesterday