SIP Passthrough

[LexLuthor]

Hi, I have an RT-AC68U running Merlin 380.59.

I just started with a SIP provider and have a Polycom VX400.

My phone won't register with the SIP provider unless I have SIP Passthrough enabled. It's my understanding, that it's preferable to run with SIP Passthrough disabled. Is that correct?

If so, any suggestions on what I need to do in order to in order to get my phone to work with SIP Passthrough disabled?

 

[RMerlin]

Hi, I have an RT-AC68U running Merlin 380.59.

I just started with a SIP provider and have a Polycom VX400.

My phone won't register with the SIP provider unless I have SIP Passthrough enabled. It's my understanding, that it's preferable to run with SIP Passthrough disabled. Is that correct?

If so, any suggestions on what I need to do in order to in order to get my phone to work with SIP Passthrough disabled?

Set the SIP Passthrough to Enabled. That will open the port, without running the NAT helper.

If it doesn't work, try again but with Enabled + NAT Helper.

 

[LexLuthor]

Set the SIP Passthrough to Enabled. That will open the port, without running the NAT helper.

If it doesn't work, try again but with Enabled + NAT Helper.

Thanks for the reply. I currently have it set to Enabled + NAT Helper and that does work. I was under the impression that disabled would be the optimal setting, but with it disabled, it doesn't register. Is disabled not where I'm ultimately trying to get it? If disabled isn't optimal or I can't get it to work that way no matter way, is Enabled better than Enabled + NAT Helper?

 

[jack901]

mine is setup as enabled and works fine. never tried nat helper setting but if it works for you ,would just use it. if I set to disabled it does not connect.

 

[LexLuthor]

Yes, same thing as me where it doesn't connect if SIP Passthrough is disabled.

What I'm trying to get at in this post is would my SIP would better/more reliably if I could get it to work with SIP Passthrough disabled? Is it better to run SIP with that option disabled as opposed to Enabled or Enabled+NAT Helper? If so, then the next step would be to figure out how to get it to work with SIP Passthrough disabled.

If it doesn't really matter, then I'd just leave it be.

 

[jack901]

I would say if it works with it enabled it does not really matter. I have had this setup for years and no issue. I think sip passthrough is enabled by default.
I get clear voice both ways .I have a grandstream behind the router and works fine. I see alot where they want the ata before the router but I have had no success with that.
if it works fine the way you have it thats all that matters.

 

[Kronyx]

It's not working with my cisco ATa, people can't ear me.

 

[RMerlin]

Thanks for the reply. I currently have it set to Enabled + NAT Helper and that does work. I was under the impression that disabled would be the optimal setting, but with it disabled, it doesn't register. Is disabled not where I'm ultimately trying to get it? If disabled isn't optimal or I can't get it to work that way no matter way, is Enabled better than Enabled + NAT Helper?

Usually, you want to use the NAT helper if you connect to a remote server, and disable it if you run your own local PBX.

 

[RMerlin]

It's not working with my cisco ATa, people can't ear me.

I don't have any problem with my Cisco SPA112. Check your ATA configuration.

 

[Kronyx]

Trust me I have checked my configuration It work behind the HH2000 so it's not the ATA configuration.

 

[RMerlin]

Trust me I have checked my configuration It work behind the HH2000 so it's not the ATA configuration.

This is the configuration that I use:

http://wiki.voip.ms/article/Cisco_SPA112

On the router side, I enable SIP passthrough, and nothing else.

 

[Kronyx]

The only difference with you is that my RT-AC3200 is straight on the Bell ONT with PPPOE enabled.

 

[dvohwinkel]

I have a RT-68U on RMerlin's 380.59 .. I don;t see the SIP passthru, only the SIP NAT helper. Where do you guys see the SIP Passtru option?

 

[RMerlin]

I have a RT-68U on RMerlin's 380.59 .. I don;t see the SIP passthru, only the SIP NAT helper. Where do you guys see the SIP Passtru option?

That setting can be set to disabled, enabled, or enabled + NAT helper.

 

[dvohwinkel]

That setting can be set to disabled, enabled, or enabled + NAT helper.

It would probably help if I actually said the right version I was on.. 380.58 not 380.59. On mine I see "Enable SIP NAT Helper" user NAT pass thru.. and it's options are "Enable or Disabled"

 

[RMerlin]

It would probably help if I actually said the right version I was on.. 380.58 not 380.59. On mine I see "Enable SIP NAT Helper" user NAT pass thru.. and it's options are "Enable or Disabled"

That was added in 380.59.

 

[TOD]

Most modern ITSP's equipment does not require the inclusion of any 'NAT-helpers'(STUN/TURN/ICE & etc.).
These functions serve only one purpose - tell ITSP's server to which address and port sending voice traffic (RTP), so that it reached the customer equipment. Pretty simple manner it is achieved on the side of telephony servers themselves. Usually the only thing that is required, activate in IP-phone or softphone - NAT Keep-Alive, to periodically sending packets to server's address, to maintain opened port on the router. Otherwise, incoming calls won't reach to client's IP-phone (since the port on the router has already closed).

I recommend to start to check registration and make calls via MicrosSIP(nice and simple softphone for Win, http://www.microsip.org/downloads) on computer. That is much easier to capture signal / voice packets through Wireshark for diagnosis.

 

[bits]

If you are turning on sip pass through you are doing it wrong or at least your client is.

You need your sip client to fill your return address within its sip messaging correctly.
If you are sending signed messages to a server to connect rtp to you on 192.168.1.103:36017 things are going to fail. A man in the middle altering signed messages will fail by design.
The client must fill it's signed messages with the correct return address.
Separately you must have the correct port forwards for your sip messaging and rtp to be able to connect.

Otherwise depend on your VoIP provider to run a session border controller than can manage the incorrect messages you send them.

Similar if you need ftp alg etc, you are doing it wrong
Stop your clients sending junk data plus use upnp, static port forwards or publicly addressable machines.

 

[Kronyx]

If you are turning on sip pass through you are doing it wrong or at least your client is.

You need your sip client to fill your return address within its sip messaging correctly.
If you are sending signed messages to a server to connect rtp to you on 192.168.1.103:36017 things are going to fail. A man in the middle altering signed messages will fail by design.
The client must fill it's signed messages with the correct return address.
Separately you must have the correct port forwards for your sip messaging and rtp to be able to connect.

Otherwise depend on your VoIP provider to run a session border controller than can manage the incorrect messages you send them.

Similar if you need ftp alg etc, you are doing it wrong
Stop your clients sending junk data plus use upnp, static port forwards or publicly addressable machines.

Should I disable the SIP passthough and redirect the ports manually ?

 

[sfx2000]

Pls don't feel bad - the problem with SIP and gateways is there are a handful of specs, and they conflict with each other.

One would think they were safe with a vertical solution (all same vendor) -

Small business had a Cisco IP-based PBX that had a SIP trunk to their telco provider (if I recall it was 8by8) - their cisco firewall/gateway was end-of-support/end-of-life and died, so they bought a replacement (newer box) - and it took their CCNE two days and multiple calls with Cisco to get things working again :|