Help diagnosing VOIP problem
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!
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?
- 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)
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.