GXE5024 and Linksys SPA921 interop

GarethM · #1 (**permalink**) April 22nd, 2009, 08:12 PM

Hi there,

A while ago I started using a Linksys SPA9000 and was just completely baffled by how many problems I had with it. I have decided to migrate to a GXE5024 to replace the SPA9000 and am so trying to migrate the config over. THE 5024 is much easier to set up and I have most of the features working so far.

I can get my SPA921 to register fine and can make internal calls through the GXE absolutely fine. The only problem is when it is involved in an external call through my SIP trunk. If my mobile calls in to the 921, then I can hear the 921 on my mobile but no audio comes back to the 921.

Some things I have tried / worked out:
- If the external caller goes through to voicemail, you can hear the audio just fine (and the message sounds fine).
- If I pick up the call on the 921 (and don't hear the audio) but then transfer it to my PSTN port on the GXE, audio gets through to the PSTN phone.
- If the external caller goes directly through to a phone connected to the PSTN ports on the GXE, the call works fine with 2 way audio.
- The audio sent back to the mobile from things like the auto-attendant etc is fine.

So this points to some kind of interop / mis-configuration between the GXE and the SPA921. Does anyone have any clue about how to diagnose this problem? The 921 config pages are huge and quite complex - but I can't think of anything that could stop this. I can't believe that i've come so close to ditching my SPA9000 only to be foiled by this!

If anyone has any clue - any clue, I'd love to hear it....
Cheers,
Gareth

Billx · #2 (**permalink**) April 23rd, 2009, 02:46 PM

It sounds like a Codec trouble.

Make sure that the phone has all the available Codecs.

711 729 etc.

Also, set your first Codec choice (in the phone) to whatever your SIP provider prefers.

Hope this helps.

GarethM · #3 (**permalink**) April 24th, 2009, 10:29 AM

Thanks for your reply - someone else mentioned it might be codec trouble. To be honest I'm not sure what that means. I've tried forcing the SPA to use different codecs in it's config and sometimes that causes the phone to not work, but others just exhibit the same problem. I don't understand why I'd be able to make internal calls just fine though - this only happens when a call comes in from the SIP trunk.

I've upgraded the 921 to the latest firmware, no change. I've changed the RTP packet size to 0.020 (which apparently helps some things) - no change.

How do I find out what codec my SIP provider is using? Surely once it hits the GXE then it takes over - or is it just passing everything straight through without touching it? In which case it should just work - I mean, it worked through my SPA9000.

Any other ideas?

Billx · #4 (**permalink**) April 24th, 2009, 02:01 PM

Thanks for your reply - someone else mentioned it might be codec trouble. To be honest I'm not sure what that means.

A codec is a means of compressing your voice so it uses less bandwidth. Popular Codecs are G711 729 etc. Its like MP3 and MP4 compress music.

---------------

I've tried forcing the SPA to use different codecs in it's config and sometimes that causes the phone to not work, but others just exhibit the same problem.

Your phone may need to use different Codecs at different times. If you lock it in on one (lets say 711) and the device you communicating with only uses 729 then your phone won't work. No voice.

----------

I don't understand why I'd be able to make internal calls just fine though - this only happens when a call comes in from the SIP trunk.

In a small network bandwidth is not a big concern.Your internal calls are most likely using G711, the widest bandwidth.
You SIP calls may rerquire that the 729 or 723 codec (less bandwidth) be used. If your phone doesn't have say 729, and your SIP Service provider uses 729 then it won't work. This is alo true for youe Grandstream PBX.

-------

I've upgraded the 921 to the latest firmware, no change. I've changed the RTP packet size to 0.020 (which apparently helps some things) - no change.

-----

How do I find out what codec my SIP provider is using?

This gets a little deep here. In the SIP Invite message there is a Session Description Protocol section. That is where you will see the codecs that are available. The phone and the SIP Service provider are supposed to negotiate a common codec to use. That is something I can't help you with.

I had the same exact problem with an Aastra 57i telephone. It worked great from station to station (on PBXNSIP) but using Callcentric.com (SIP Service) it would fail every time. Callcentric had me add 729 codec to phones list of choices and it works OK now.

-----

Surely once it hits the GXE then it takes over - or is it just passing everything straight through without touching it? In which case it should just work - I mean, it worked through my SPA9000.

I think the Grandstream may either pass through the packets or do Transcoding from one Codec to another.

----
Any other ideas?

Let us all know if any of this helped and what (if any) resolved the problem.

GarethM · #5 (**permalink**) May 4th, 2009, 09:03 AM

Well after much playing around I have made the following discoveries:

Firstly, I got a Grandstream Budgetone 200 and tried that in my system and that did exactly the same thing as the SPA921. So that meant it was obviously not actually a problem with the 921 config, it was a more general problem with the whole setup. The problem as it turns out is that I do not want to use the GXE as my network router. As such, I had the LAN port configured with a fixed IP address on our network and I configured the WAN port with similar settings (different IP address of course) but didn't actually plug anything in it. The GXE operates perfectly in this mode, with the one exception that you can't hear voice from external callers phoning in from a SIP trunk (which is a bit of a show-stopper!)

This was proved when I then tried using the GXE as the router (i.e. cable modem in the WAN port, local network in the LAN port) and what do you know all the voice was working fine. I messed around with my network settings, disabled all firewalls - even tried using it on my home internet connection (which is a very simple netgear wireless router plugged into a cable modem) still no audio. It seems that if you have the GXE behind a NAT'ing router, you loose audio from SIP trunk calls.

So my question is this - can the GXE be used behind a router in the way I described? I don't really want to use the GXE as my router - I have a much better one I'd prefer to use. It doesn't make sense that it would all work apart from inbound audio. Does this sound like a bug??

GarethM · #6 (**permalink**) May 13th, 2009, 04:57 PM

I finally worked out what was going on here. The fix to make this all work was to use the WAN port only - and not the LAN port at all when you put the GXE behind a NAT'ing firewall.

However, there is one important thing that other users should be aware of that really caught me out.

If the WAN and LAN port are set to the same subnet (even if you are only using one port) - for example 192.168.1.20 and 192.168.1.21 then the WAN port will not work correctly at all behind a NAT (i.e. not using the GXE as a router). This is what I got stung with. The way to do it is to set your WAN address to whatever you want the GXE to be on your local network and then set your LAN IP address to something random - it doesn't matter what it is as long as it is not in the same subnet as the WAN address. So something like 192.168.2.20 will be fine (if your network is using the 192.168.1.x network).

Then simply plug in the WAN cable ONLY to your network - and everything works fine! Glad I got there in the end. Thanks all for your help.