My SIP based Android app doesn't work with my RT-N66U

[RMerlin]

I found this in another forum:

<snip>
For the Asus RT-N56U there is only one way of disable the SIP ALG function with the latest firmware.
Activate the telnet function of the router and do it by hand.

nvram set nf_sip=0
nvram commit
reboot

That's it the SIP ALG should be disable.

This should solve alot of head ache. Not only sip registration.
</snip>

Is this something I could try? Not sure how to do telnet on the router though.

That information is outdated. Asus has since added the SIP Passthrough option that you already saw - this is what this option is about.

 

[cdysthe]

I'm getting 50 mbps Suddenlink cable Monday. I'm sure I'lll be back with new gripes and problems, but this one I consider closed as far as I am concerned. Thanks y'all!

 

[MotuXT]

Hi Merlin,

Thanks for your amazing firmware. I've been having some trouble with both Groundwire and Bria VOIP clients on my iPhone with my work VOIP provider. After a variable amount of time (20 secs to 20 minutes) I lose the ability to hear the person on the other end. Both apps worked fine with my previous Apple Airport router and also work fine over 3G.

I note that hear you say "SIP passthrough must be disabled when using a client. That option is for when you are running a SIP server (for instance, Asterisk)."

However, in this thread you say:

"Usually, you want this enabled if you use a SIP phone to connect to a remote PBX, and disabled if you host your own PBX server at home (Asterisk or other)."

I'm a bit confused as to which it is?

Thanks!

Alex

 

[RMerlin]

Hi Merlin,

Thanks for your amazing firmware. I've been having some trouble with both Groundwire and Bria VOIP clients on my iPhone with my work VOIP provider. After a variable amount of time (20 secs to 20 minutes) I lose the ability to hear the person on the other end. Both apps worked fine with my previous Apple Airport router and also work fine over 3G.

I note that hear you say "SIP passthrough must be disabled when using a client. That option is for when you are running a SIP server (for instance, Asterisk)."

However, in this thread you say:

"Usually, you want this enabled if you use a SIP phone to connect to a remote PBX, and disabled if you host your own PBX server at home (Asterisk or other)."

I'm a bit confused as to which it is?

Thanks!

Alex

My bad. The original post was correct - the helper is needed for clients (I'll amend my previous answer to avoid any further confusion).

EDIT: now I see why the confusion. While the option is called "passthrough", it's actually a NAT helper. You need the helper for clients, while you want it Disabled if you have a server.

 

[MotuXT]

Ahh ok. Strangely I ran a 10 minute test with it disabled last night that worked without the problem happening... as it happens intermittently I might have just got lucky though. It is really strange for the call to go through fine and then sometimes just suddenly lose the audio of the person on the other side. It makes me think the router suddenly stops passing through the packets. Do you think the NAT helper could do that?

 

[RMerlin]

Ahh ok. Strangely I ran a 10 minute test with it disabled last night that worked without the problem happening... as it happens intermittently I might have just got lucky though. It is really strange for the call to go through fine and then sometimes just suddenly lose the audio of the person on the other side. It makes me think the router suddenly stops passing through the packets. Do you think the NAT helper could do that?

Not sure.

I recommend you check with the voip.ms Wiki. While they might not be the provider that you use (that's the one I use personally), they have very good documentation on how to configure different devices/clients with their service. Since they use a very standard Asterisk server, there's a good chance that their recommendations might also apply to your provider.

 

[MotuXT]

OK, thanks for the info.