UDP connections drop out for VoIP phones

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bapujidaladla

1 post

Wannabe Geek

#310659 10-Nov-2023 11:32

Hi Team,
We have clients with 2 different Internet connections running through MikroTik routers. What we have noticed is happening for these clients, that the phones will start dropping out the connections for phones on UDP protocol. Session timeout even is setup for 3600 in the phone will only be staying under 10. We feel something is forcing the connection to only have a session timeout of 10 seconds or below and then will drop the connection. Problem will be resolved if we change the protocol to TCP, but we want to stick to UDP for VoIP for better communications. Can you please suggest any troubleshooting for this issue?

darylblake

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#3158001 10-Nov-2023 12:51

Most likely they are on fibre, and the SBC would be somewhat very close to (a few ms away) so I don't suspect its a packet loss issue.

Have you tried using other phones? Different models... I suspect that is the common denominator here? I suspect an issue with the handset itself?

MadEngineer

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#3158006 10-Nov-2023 13:11

Have you got the helper enabled? Do your VoIP devices require it?

You're not on Atlantis anymore, Duncan Idaho.

speed

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#3158063 10-Nov-2023 14:44

Enable 'SIP options' keep-alives in the PBX configuration and set the interval to less than 10 seconds.

To be honest though, if your firewall's NAT timeouts are so aggressive that your ports are only open for 10 seconds post last-data then you might as well just move to TCP. You shouldn't see any real-world degradation of quality using TCP as long as the latency between your PBX and phones isn't too great and you're not being affected by constant unreasonable (eg >1%) packet loss.

RunningMan

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#3158069 10-Nov-2023 15:03

In the Mikrotik

IP/Firewall/Service Ports and enable the sip helper. Default is 1 hour.

MadEngineer

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#3158130 10-Nov-2023 21:55

Don't blindly go enabling or disabling it however. You need to check if the devices your using require it, many require it turned off.

You're not on Atlantis anymore, Duncan Idaho.

muppet

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#3158224 11-Nov-2023 10:40

Just change to TCP, that'll just be signalling (SIP)

The actual voice traffic will still be RTP, which is UDP.

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RunningMan

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#3158230 11-Nov-2023 10:54

MadEngineer:

Don't blindly go enabling or disabling it however. You need to check if the devices your using require it, many require it turned off.

From memory the default is enabled, but the time may need to be tweaked.

MadEngineer

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#3158434 11-Nov-2023 22:40

Is it running on port 5060 or 5061?

You're not on Atlantis anymore, Duncan Idaho.

raytaylor

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#3158454 12-Nov-2023 09:17

If you change your sip transport to TCP that will help with registration, but the audio is still a separate UDP stream.

Things to consider

How does your router perform the load balancing?
Sip registration should be every 5 minutes however something to note is that if your randomly registering via a different upstream each time it would cause problems
I would suggest setting rules that send voip out one upstream and have that switch to another only as a failover rather than load balancing
That could be achieved with a static /32 route with monitoring enabled
Assign RTP audio ports for each voip phone within the local network and set up the appropriate UDP port forward rules in the router.
You dont need to do sip rules for each phone if you are using TCP for sip transport

Ray Taylor

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