Intermittent Audio Issues

I've been fighting this issue for over 27 hours. When dialing sometimes audio is not hear on the receiving end and other times audio is not heard on the calling end. It appears audio is intermittent at best thus far in my setup. I'm running astersk and using VOIPo as a trunk. I also have several other SIP carriers in which I do not experience this issue at all. It seems VOIPo may be transmitting audio on ports I have yet to define within my firewall. Any insight as to the ports VOIPo is using beyond the standard 5060 SIP/10,000-20,000 RTP?

Thanks,

Chris

I found a snip from another post:

"I think VOIPo suggests udp 5004-65000"

Now why on earth would almost 60,000 ports have to be opened for a SIP service provider? Even though I only pass specific IP's through my firewall this is a large and I mean a large security risk for all of us. I would like to narrow down this list just a tad bit and not open up almost 60,000 ports on my firewall for a simple SIP service provider. Does anyone have a narrow list of ports for VOIPo? I'd hate to have to resort to a packet analyzer and "hope" I am able to capture all the dynamic ports.

Opened up the recommended, almost, 60,000 ports and am still having audio issues. I also have noticed from time-to-time when dialing my VOIPo number...it has a fast busy and sometimes I cannot hear ringing when dialing a number. The number should go to voice mail not a fast busy.


Chris

I have yet to resolve this intermittent audio issue. One call it's there....three calls later it isn't then it's back the fourth, the third, the eight? No consistency at all. Not sure what the issue is however it appears to be on VOIPo's side due to the fact sometimes the number has a fast busy and refuses to go to their voice mail system. Very inconsitent service at best on their BYOD side of the house. I'll have to contact their support dept tomorrow.

If you have an ata or softphone you might want to make sure the line works correctly there before setting it up in asterisk. I have also never needed to forward ports for my service to work. I use 2 Obi110s with unique sip and rtp port ranges configured on the Obis - no config on the firewall is needed.

Quote:

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Originally Posted by chrisb009

Even though I only pass specific IP's through my firewall this is a large and I mean a large security risk for all of us.

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1) It's only a security risk if something is listening on the permitted ports.
2) You configure what incoming ports Asterisk will negotiate during the call setup by defining rtpstart & rtpend in the RTP config. You only need to open those ports through your firewall.

One-way audio is almost always a NAT mapping failure, so start troubleshooting there.