VoIP One-way Audio and Voice drops
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@scottalanmiller said:
- Did anything change on the firewall?
Nothing had changed on the firewall end that I know of. The last update was on Monday of the week before.
- Is SIP-ALG (sometimes under a different name) getting enabled somehow?
I don't see an option for SIP-ALG on the Meraki firewall? Is this where this would be setup or would it be on the PBX itself? I do have SIP (Voice) traffic and traffic destined for the SIP trunk setup with a higher priority then all other traffic in the network. This has been enabled since we started with the FreePBX server last November.
- Did your STUN server (an external thing) go down during that time?
Where would the STUN server be located? Is it generally something that is selected or does it go through the ISP?
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@scottalanmiller said:
Things like latency pretty much cannot cause one way audio. Latency and packet drops do not manifest in this manner.
That is good to know. The odd relationship between latency and voices drop outs was purely coincidence. If the router's queue was overloaded would you see this behavior or would it be more robotic audio and lagged communications?
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I tried a SIP-ALG detector from Nextiva (http://dc.nextiva.com/sipalg/). It says there is no SIP-ALG.
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@coliver said:
Where would the STUN server be located? Is it generally something that is selected or does it go through the ISP?
Generally this is set on the phones and the PBX.
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@coliver said:
@scottalanmiller said:
Things like latency pretty much cannot cause one way audio. Latency and packet drops do not manifest in this manner.
That is good to know. The odd relationship between latency and voices drop outs was purely coincidence. If the router's queue was overloaded would you see this behavior or would it be more robotic audio and lagged communications?
That does suggest that the router was the issue and having problems. When fully loaded it might have behaved badly.
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Sounds like it would either be a network issue or the SIP Provider.
Meraki APs are great, there routers not so much but That may not be your issues unless it's undersized for your company.
Re-tramsmit suggests to me it could be a MTU issues or a QoS issue (mis configuration or under powered router)
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@thecreativeone91 said:
Sounds like it would either be a network issue or the SIP Provider.
Meraki APs are great, there routers not so much but That may not be your issues unless it's undersized for your company.
Re-tramsmit suggests to me it could be a MTU issues or a QoS issue (mis configuration or under powered router)
I was on the phone with Meraki support for several hours on Friday. I've been having an on going issue with their malware protection option that has caused the router to lockup at 100% cpu usage (which you have no way of knowing because their GUI is way too simple), which I've disabled. The support rep was confident that it wasn't a capacity issue.
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@coliver said:
@thecreativeone91 said:
Sounds like it would either be a network issue or the SIP Provider.
Meraki APs are great, there routers not so much but That may not be your issues unless it's undersized for your company.
Re-tramsmit suggests to me it could be a MTU issues or a QoS issue (mis configuration or under powered router)
I was on the phone with Meraki support for several hours on Friday. I've been having an on going issue with their malware protection option that has caused the router to lockup at 100% cpu usage (which you have no way of knowing because their GUI is way too simple), which I've disabled. The support rep was confident that it wasn't a capacity issue.
I wonder how their support rep was so confident in it not being a capacity issue? Seems rather unlikely that it was, but Meraki is not high end gear and it could easily be. That there was an issue like that around the same time is awfully suspicious.
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@scottalanmiller said:
@coliver said:
Where would the STUN server be located? Is it generally something that is selected or does it go through the ISP?
Generally this is set on the phones and the PBX.
Does this still apply if the phones and PBX are on the same network? All phones and registered extensions are local to this site.
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@coliver said:
@scottalanmiller said:
@coliver said:
Where would the STUN server be located? Is it generally something that is selected or does it go through the ISP?
Generally this is set on the phones and the PBX.
Does this still apply if the phones and PBX are on the same network? All phones and registered extensions are local to this site.
It applies to the PBX but not the phones, in that case.
Are you getting one way audio on extension to extension calls too? Or only when hitting the external trunk?
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@scottalanmiller said:
@coliver said:
@scottalanmiller said:
@coliver said:
Where would the STUN server be located? Is it generally something that is selected or does it go through the ISP?
Generally this is set on the phones and the PBX.
Does this still apply if the phones and PBX are on the same network? All phones and registered extensions are local to this site.
It applies to the PBX but not the phones, in that case.
Are you getting one way audio on extension to extension calls too? Or only when hitting the external trunk?
Only when hitting the external trunk. The one way calling was happening on every external call either people calling in or extensions dialing out.
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Yeah, that is almost certainly an issue with the Meraki and/or missing STUN settings. It is possible that under light load and no STUN, things "just worked" and only when things get heavy that STUN is needed. That's one option. Another is that the Meraki simply behaved badly under load.
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Experiencing the same issues again this morning. I get some flapping from my SIP trunk provider but not on every call.
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I hope @coliver doesn't mind me asking a few VOIP questions here considering his issues.
@scottalanmiller mentioned STUN servers. In the case of Coliver where he only has internal phones, is the STUN server provided by his SIP provider? or does Coliver need to set one up himself?
If Coliver wanted to have phones outside of the network, would Coliver have to set a STUN server up himself? Am I correct that is seems that a STUN server sorta acts like a reverse proxy for SIP?
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@Dashrender said:
I hope @coliver doesn't mind me asking a few VOIP questions here considering his issues.
@scottalanmiller mentioned STUN servers. In the case of Coliver where he only has internal phones, is the STUN server provided by his SIP provider? or does Coliver need to set one up himself?
If Coliver wanted to have phones outside of the network, would Coliver have to set a STUN server up himself? Am I correct that is seems that a STUN server sorta acts like a reverse proxy for SIP?
Nope, no problems I was wondering the same thing myself.
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Just got a report that users who use our Network Extender are also having the same issues that our "landline" users do.
The network extender, from my understanding, is basically a VoIP gateway. Which tells me I can rule out both the PBX and the SIP Trunk if a different version of VoIP is also having issues.
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@Dashrender said:
I hope @coliver doesn't mind me asking a few VOIP questions here considering his issues.
@scottalanmiller mentioned STUN servers. In the case of Coliver where he only has internal phones, is the STUN server provided by his SIP provider? or does Coliver need to set one up himself?
There is no STUN server in this scenario. STUN is for devices to PBX, not SIP trunks.
If Coliver wanted to have phones outside of the network, would Coliver have to set a STUN server up himself? Am I correct that is seems that a STUN server sorta acts like a reverse proxy for SIP?
You are not required to setup a STUN server for outside phones, but it can smooth out problems when dealing with end users behind gear that may or may not play nice. For @Bundy I use a public STUN server simply because there are only 2-3 extensions that are not inside an OpenVPN tunnel in the first place.
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@JaredBusch said:
For @Bundy I use a public STUN server simply because there are only 2-3 extensions that are not inside an OpenVPN tunnel in the first place.
Public STUN server? can you give more details?
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I setup a secondary trunk via Vitelity, attached it to the PBX and called in and out. This caused the same issues we were seeing previously. I then attached that trunk to my handset locally, bypassing the PBX, this also caused the same issue we have been seeing.
I've ruled out the PBX and SIP Trunk... all that remains is our firewall/router and our internet connection.
