Voice is lost in both direction after 2:30 min

Hello everyone.

I have very strange problem and I can’t manage to find the reason by myself, so I rely on your experience. Everytime my extensions are calling, after exactly 2:30 minutes (150sec) the voice is disappearing in both directions, but call session is not disconnecting, they only are unable to hear each other. I have run debug mode, but there is no error rising during the losing the voice :frowning_face: I have tried asterisk -RvvvvvT The only warning that I’m seeing is this one, but this one is rising just after call is starting:

WARNING[44206][C-00000b90]: translate.c:407 framein: no samples for ulawtolin

Then call continues exactly 2:30 minutes and after is silently lost, but the timer on Phones are continue ticking and there’s no hung up signals, so call continues without voice. Here I should mention, that voice is lost only if the call is going via the trunk i.e. either outbound or inbound calls. If the call is between extensions, there is no such problem.

FreePBX version is, I have upgraded to the latest build version - 10.13.66-22, I’m using Asterisk version - 13.19.1, all peers are using Chan_Sip, Enabled codecs are: ulaw, alaw, gsm and g726. RTP port range is set to 10000-20000.

Does anyone have the idea, what might cause such behavior? Hope that somone can give me the clue where should I search the reason. The number 150 sec makes me think, that there is some king of parameter to turn off RTP traffic after this, but where is it?

Thanks everyone in advance.

IMO this is likely a networking problem caused by external hardware (firewall, router, modem, etc.)

You can confirm this by capturing traffic at the PBX with tcpdump. Start a capture and make a test call. Wait until voice is lost, then end the call and stop the capture. Move the pcap file to your PC and examine it with Wireshark. In the early part of the call, you should see four RTP packets every 20 milliseconds; from extension to PBX, from PBX to trunk, from trunk to PBX, and from PBX to extension.

I suspect that when the voice fails, you will still see RTP going to the trunk but the remote party no longer hears it, which would confirm that another network device is causing the trouble.

I assume that the PBX and extensions are at the same location. If not, please explain (cloud system, branch office, etc.)

Is this a new system that never worked properly, or one that just started to fail? If the latter, are you aware of any change that may be related?

Please post: ISP? Type of connection (fiber, cable, DSL, LTE)? Modem make/model? Separate router, if any? Separate hardware firewall, if any? VoIP-aware switch? Using VPN? Using VLAN? Is PBX server physical or virtual? If Virtual, which platform? Phones make/model? Trunking provider? Any other relevant info about your environment?

Hi Stewart1, thanks for your reply and suggestions. We have managed to find the reason, which seems quite strange to me, but the fact is that it helped indeed.

I forgot to say that FreePBX is a Vritual Server based on VMWare platform. The system is running about 3 years, maby more, I just periodically upgrading it via the upgrade scripts. We have extensions on local subnet and over VPN as well.

So, I will write down solution in our case. The only thing we had changed recently is that, we have changed ISP. Earlier Internet service provider and VoIP Service provider was the same company, but now for some reasons, we have changed an ISP, but we get VoIP service from the same company as we used to. So, when i tried everything, that I could, I called to them (VoIP provider) and they say, that the problem seems to be codec. I explain them that I didn’t change anything on server and sent them the screen of enabled codecs. It was: ulaw, alaw, gsm, g726. They said that i should disabled all codecs except of alaw. I always thought that this list means, that this is the codecs which system could talk and then during the SIP session, the peers are negotiating the correct one, which both side could use. I also thought that if codecs doesn’t match, systems could not talk to each other at all, but in my case, the voice traffic worked, but was disconnected after 2:30 minutes. But VoIP provider insisted to remove the odd ones. So I left only alaw and disable others and it actually helped.

So, if anyone will have the same problem, keep in mind, that it could be caused by codecs.

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