VoIP calls dial out but unable to hear on Voice VLAN

Obiwan_3011
Just browsing

VoIP calls dial out but unable to hear on Voice VLAN

We have an MX95 and two vlans.  One is the default vlan of 1 and our voice vlan is 10.  VoIP calls are working fine internally, however whenever we attempt to dial out, it works but we are unable to hear anything.  I have reviewed our rules from our old ASA and everything looks correct.  Not sure if it is a routing issue or not.   Looking for some thoughts or ideas if anyone has experienced this and what suggestions do you have to help solve it.   Everything else on the MX95 is working fine.

9 Replies 9
DarrenOC
Kind of a big deal
Kind of a big deal

One way audio is pretty much 99.99% a routing issue.  If you can’t hear the caller then you’re not receiving voice packets inbound.  Do you have a syslog setup so you can see if anything’s being dropped?  NAT rule inbound for voice traffic?

Darren OConnor | doconnor@resalire.co.uk
https://www.linkedin.com/in/darrenoconnor/

I'm not an employee of Cisco/Meraki. My posts are based on Meraki best practice and what has worked for me in the field.
Obiwan_3011
Just browsing

We do have a syslog server; I will forward to that.   Do you have an example of the inbound NAT rule to forward to the correct vlan for voice traffic.   Just looking to compare something against.  Thanks.  I believe it is also a routing issue but trying to nail down the specific issue.

alemabrahao
Kind of a big deal
Kind of a big deal

There are some tips.

 

VoIP on Cisco Meraki: F.A.Q. and Troubleshooting Tips - Cisco Meraki Documentation

I am not a Cisco Meraki employee. My suggestions are based on documentation of Meraki best practices and day-to-day experience.

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PhilipDAth
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Kind of a big deal

Are there any missing inbound NATs compared tot he old ASA?

 

Are you using SIP registration or SIP trunking?

Obiwan_3011
Just browsing

We are using a SIP registration.

PhilipDAth
Kind of a big deal
Kind of a big deal

The best way I have found to solve most of these is to use SIP-TLS (as opposed to SIP-UDP).  Failing that SIP-TCP.

 

Are you able to use any of those options?

IvanJukic
Meraki Employee
Meraki Employee

Hi @Obiwan_3011 ,


If you dial in, does that work or the same? If calls connects ok and you get silence. Most likely the RTP streams are being blocked or dropped. So, double check for things like (1:1 NAT, port forwarding, etc) on the Firewall. 

Also where is the PBX/Platform hosted locally, remote site or in the Cloud? 


Cheers,

Ivan Jukić,
Meraki APJC

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Obiwan_3011
Just browsing

Dial in behaves the same way.   Call connects but we get silence.   The platform is hosted locally.   I believe we have that setup correctly for the RTP streams, the only difference I can see is that the voice is a separate VLAN.   Do you have any examples of how that should look on the Meraki side?

IvanJukic
Meraki Employee
Meraki Employee

 

If it is happening Inbound too. Then that would mean that the problem is 'No Voice Path'. I.e. RTP/Audio streams. This is mostly due to a problem with signalling or due to Firewall/NAT/Port Forward issues.

 

Check the Voice Platform for indications of signalling problems. Check the configuration of firewall, switch, routers etc on the voice path. If you only have 2 VLANs, you might want to use 1 VLAN for Data and the other for ALL VoIP related traffic. Of course, always consult the Vendor's Platform guide. See below Meraki guide for common tips.

 

https://documentation.meraki.com/General_Administration/Tools_and_Troubleshooting/VoIP_on_Cisco_Mera...

 

And remember, Packet Captures are your best friend. 😁


Cheers,

Ivan Jukić,
Meraki APJC

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