Asterisk SIP One way Audio

ok, I know what your all thinking just from the title alone.
It’s SIP and One way Audio, it’s going to be a NAT and firewall issue. That was my immediate thought too, but it’s wrong.
To explain a little of the background will probably help.

I’ve been running Asterisk at home now for years, using a Digium 422. One line coming in, and a few Linksys SIP phones.
I’ve been happy with it running away for some time, adding in blacklist functions into the dial plan of incoming calls, blocking withheld numbers into entering their number and building up a list of numbers that are annoying me to reject. I’ve had a few people questioning when they’ve called about it. and even more from visitors whose witnessed the Tivo auto pausing if a call comes in as well as displaying who it is.

So after years of me singing it’s praises (as well as a linux server in general), it’s no-wonder that someone else wanted me to do a similar setup. Enter cousin 🙂

After weeks of searching on ebay, finally someone is selling a digium card. So bought it up quickly.

He’s had a Linux server running for a good few months, and I’ve got it all VPN’d together with my network.
So I installed the Digium card, installed asterisk, and went about configuring it using my config as the basis.
It all worked well, incoming and outgoing calls all fine using the card 2 phones connected and one line.
But obviously, now you’ve got a little you want more. So off he went looking at phones, I have to be fair he found a great deal of linksys phones, 4 of them with power for £40 when most are selling single units at over £20.

So I spend another few hours running network cables to 4 different locations in the house, connect up the phones and go back to configuring asterisk and the phones. Fairly straight forward, run through testing phone to phone etc, all appeared fine.

The following day I had a lovely call saying that their now getting one way audio. Straight away I thought that’s NAT, but as all the phones and asterisk box are on the inside NAT shouldn’t come into it. Still I went back through the SIP config and adjusted lots of it to completely wipe out NAT (I use NAT in my config as I have some connections from SIP providers etc).
More testing, but still the same.
So I started running SIP and RTP traces, I’ve had to read through them bother before and can just about follow what’s going on, but I mainly just look for anything obvious out of place, like IP addresses that are external etc.
Still after going through them, it all looked good in the SIP debug, but the RTP was only showing one side audio on calls that were going in / out over the Line. Calls to other phones were fine.
So I registered one of the phones onto my system and placed a call, that worked fine.
So I come home and registered one of my phones to his system. Placed the call, and again one way audio.

I’m not convinced it’s a NAT problem as my phone would now be running over the VPN to his server and it’s the same server that terminates the VPN. I now started checking out the codecs, but testing using different ones just pointed to it not being codecs. If it couldn’t find common ground the call failed from the start.

It took a few days to get back to as I was onto other stuff, but check after check just showed that everything was fine. But still the one way audio on calls.

I looked again and there was a new version of asterisk released 10.1.3, where I was using 10.1.2 on his server. The 2 fixes in the change were around SIP. So I downloaded the new version, compiled it and installed. Made sure it was all running and then got home to make some test calls.

Still one way audio. So I’m now left with one option. My own server is running an older version of asterisk, I uninstalled his asterisk completely and then downloaded the source for, compiled and then installed.
Using the same config from the 10.1.3 install threw a few warnings on loading, but it still loaded. Got him to make a test call, he called my mobile and hey presto we can hear each other.

I wanted to update my own asterisk, but after encountering this issue and especially as it’s still present in the latest I think I’ll leave updating my system for a while.

Bottom line, dont just consider One way Audio a NAT problem, and if everything looks to be working correct consider downgrading if there’s no upgrade available.

2 thoughts on “Asterisk SIP One way Audio”

  1. Firstly apologies it took me so long to publish your comment (about a month) I didn’t know it was there 🙂
    I appreciate the use of SIP providers, I personally prefer IAX I find them far more stable. Unfortunately it leaves a system dependant on internet, also I have to pay for a phoneline so may as well use it hence the Digium. If I didn’t have to have a phoneline for ADSL/FTTC I’d get rid of it and just use VoIP.

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