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cosentustech
cosentustech
New Member
September 13, 2022
Question

No audio on internal calls

  • September 13, 2022
  • 2 replies
  • 10930 views

Multi-site setup. Main site has an ASA 5506, site b has a 60F, site C has an ASA 5506.

Ever since adding the 60F into the mix we have had problems with internal calling. All sites used to be 5506.

The phone server is at the Main site.

IPSec tunnels are built between all sites.

At Site B regular inbound/outbound calling works and extension calls to the Main site work.

Extension calls at Site B do not work, there is no audio. A user at Site B dials another user at Site B the phone rings but there is no audio either way.

The same problem happens when Site B tries to call a user at Site C, phone rings, no audio.

I tried disabling all SIP inspection on the 60F but that did not help.

Created a basic traffic shaping policy for RTP but that also did not help.

What am I missing?

 

2 replies

gfleming
Staff
gfleming
Staff
September 13, 2022

There's typically two sessions you need allow access for on a VOIP call: the setup and the payload. The setup is typically done using something like SIP (UDP 5060) and the RTP payload is typically UDP high random ports. Are you by any chance only allowing SIP UDP 5060 on your policy and not the other traffic?

 

It's also odd that calls between phones at Site B do not work. Are they on the same subnet the two phones? If so it's unlikely the firewall is getting in the way there...

    JamyBalys
    JamyBalys
    New Member
    January 25, 2023

    One thing you can try is to check if the 60F firewall is blocking the RTP traffic. RTP is the protocol used for carrying audio in VoIP calls. Make sure the 60F firewall is configured to allow RTP traffic between all sites.
    Another thing you can check is if the problem is with the phone server. Make sure the phone server is configured correctly and that it can reach all the sites.
    It's worth noting that free calls service like freetring.com may be an alternative solution for you as well.
    Let me know if you need more help or have any other questions.

    cosentustech
    cosentustechAuthor
    New Member
    September 13, 2022

     

    Thanks for the input.

     

    The tunnel policies are set to allow all traffic so in theory whether it is SIP or RTP it should be going through. And the LAN to LAN policy is set to allow all as well.

     

    All the phones at site b are on the same subnet but they do have to reach out to the phone server at the main site to complete the call. I confirmed this with a packet capture. When I dial an extension at site b I see packets from the phone I'm using hit the phone server and then packets from the phone server go back to the phone I'm trying to call. When I pick up the call there is no more traffic to see. It's really strange.

     

    I'm currently working with the phone vendor on this as well to see if some call mapping is miss-configured on their end.

     

    Thanks

    Josh

    gfleming
    Staff
    gfleming
    Staff
    September 13, 2022

    Yeah that sounds really strange. Normal for setup traffic to go to the server but RTP should be from phone to phone. Can you do a packet capture at the switchport of the phone at site B and see if it's sending any RTP packets at all?

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