Navigation

    Netgate Discussion Forum
    • Register
    • Login
    • Search
    • Categories
    • Recent
    • Tags
    • Popular
    • Users
    • Search

    Help diagnosing VOIP problem

    Traffic Shaping
    2
    4
    2072
    Loading More Posts
    • Oldest to Newest
    • Newest to Oldest
    • Most Votes
    Reply
    • Reply as topic
    Log in to reply
    This topic has been deleted. Only users with topic management privileges can see it.
    • S
      Spinmaster last edited by

      Hi Everyone,

      I'm having an intermitant fault with my VOIP service.

      Ocassionally (could be after just one or two minutes, but could be after 45+ minutes) I can no longer hear the other party while making a telephone call (sounds like a very bad mobile call).  The other party can hear me fine at all times.

      I have got…

      • PFSense v1.2
      • Local LAN + DMZ
      • Traffice Shaper turned on (on the LAN interface, where my VOIP converter is located), with only rules set up for VOIP

      I've tried looking at the Queue Status graphs while being on the phone, and have noticed that both inbound and outbound VOIP queues seem to go up to around 25Kb/s and that the graphs go no more than half way across the screen.

      What should I look at on PFSense to determine whether the problem is caused by PFSense or elsewhere?  Is there a log file that might give me some help?

      Is there a tweak that I can do to PFSense that might fix the problem?

      Thanks for any help!
      James.

      1 Reply Last reply Reply Quote 0
      • S
        stechnique last edited by

        How many VoIP phones in your house/office?
        Did you forward SIP and RTP ports to your phone with NAT? Do you have Asterisk running locally or is it a single VoIP line? Who is your provider?

        1 Reply Last reply Reply Quote 0
        • S
          Spinmaster last edited by

          Hi,

          • I've got a single telephone
          • I've got TCP port 5060 fowarded from WAN to my VOIP box (a Linksys PAP2 box)
          • I live in Australia - use a provider called Go Talk (different provider to my ADSL service - AANet)

          James.

          1 Reply Last reply Reply Quote 0
          • S
            stechnique last edited by

            SIP registration usually uses UDP 5060 and not TCP, though it could be TCP… you should double-check to make sure. You should ask your provider which RTP ports they use and forward those as well.
            One way audio in SIP calls usually has to do with RTP ports either being full (no more left to assign), or the RTP packets could get lost (improper forwarding), packet loss, etc.

            1 Reply Last reply Reply Quote 0
            • First post
              Last post

            Products

            • Platform Overview
            • TNSR
            • pfSense
            • Appliances

            Services

            • Training
            • Professional Services

            Support

            • Subscription Plans
            • Contact Support
            • Product Lifecycle
            • Documentation

            News

            • Media Coverage
            • Press
            • Events

            Resources

            • Blog
            • FAQ
            • Find a Partner
            • Resource Library
            • Security Information

            Company

            • About Us
            • Careers
            • Partners
            • Contact Us
            • Legal
            Our Mission

            We provide leading-edge network security at a fair price - regardless of organizational size or network sophistication. We believe that an open-source security model offers disruptive pricing along with the agility required to quickly address emerging threats.

            Subscribe to our Newsletter

            Product information, software announcements, and special offers. See our newsletter archive to sign up for future newsletters and to read past announcements.

            © 2021 Rubicon Communications, LLC | Privacy Policy