SIP Helper

[Authority]

What's the status of the SIP passthrough and "NAT Helper"? Does it work? My VoIP provider says they generally are broken and don't help and should be turned off?

 

[RMerlin]

It works, but whether your specific scenario will work properly with it or not depends on your service. You have to test it to determine if it works better with or without it for you.

 

[dvohwinkel]

What's the status of the SIP passthrough and "NAT Helper"? Does it work? My VoIP provider says they generally are broken and don't help and should be turned off?

I use it for VOIP.ms and it works great.

 

[Authority]

It works, but whether your specific scenario will work properly with it or not depends on your service. You have to test it to determine if it works better with or without it for you.

My provider said to turn it off , which we did, and it seems to offer no difference.

If "SIP Passthrough" isn't required for passing through SIP, it seems at the very least poorly named.


Sent from my iPhone using Tapatalk

 

[TOD]

I just tested SIP Passthrough option on RT-N66U with FW 380.62.
In all three situations there was not detected any modification in the SIP traffic or deny through the router.
It looks weird.

RMerlin
May be in this FW version, this function does not work(does not analyze the corresponding type of traffic, 5060 UDP/TCP), or only for this model router?

 

[L&LD]

Did you do a reboot of the router and associated clients after each change/test?

 

[Authority]

Did you do a reboot of the router and associated clients after each change/test?

I rebooted a couple (not all) devices. I still don't get it. If SIP devices work fine with or without what's the point of "SIP Passthrough"? Is it basically a legacy thing?


Sent from my iPhone using Tapatalk

 

[Nerre]

I cannot get incoming calls to work with any of the three settings, even when I port forward the relevant ports to the SIP dect base...

Outgoing calls work fine (audio both ways too).

SIP provider support only tells me to try to connect the dect base without the router.... they don't officially support third party devices.

 

[TOD]

I have no problems with making or receiving calls on my IP-phone at home, which works through two NAT(RT-N16 with j9527 fork and RT-N66U with ASUS-Merlin).
The problem isn't in routers, but in IP-phone's configuration and/or functionality of ITSP's VoIP-platform.

In most such cases(if IP-phone registering on the server side) problem with incoming calls should be solved by enabling the 'NAT Keep-Alive'(the names vary from manufacturer to manufacturer, but the meaning is the same) function in IP-phone configuration with sending Keep-Alive packets every 30 seconds.

Here is an example configure this feature on the phones Yealink and Gigaset:

 

[Nerre]

I already have registration refresh time set to 90 seconds and NAT refresh to 20 seconds...

And port forwarding (virtual server) pointing both 5060 and the RTP ports to the DECT base... even for both tcp and udp (while I guess udp would be enough for the RTP ports).

But I gave up years ago, everybody in the family already had a cellphone so the SIP phone wasn't used much for incoming calls anyway. I just hoped that when I upgraded to the latest version a couple of weeks ago things would maybe start working... But nope.

 

[TOD]

You cannot use port forwarding for incoming requests when using outbound registration on VoIP-server from IP-phone.
It will not affect the current problem or make it worse.
Reducing registration time make sense only when your external IP change often(in mobile network), in static network it's useless.

Delete all port forward rules, which you create for IP-phone and check 'UDP Timeout: Unreplied' value in 'Tools' >> 'Othe settings'.
The value must be no less 30 seconds.

Then test incoming call with softphone(MicroSIP for example) on your computer from the same LAN.

 

[Nerre]

As I said, I tried without forwarding rules too.

I have tried:
SIP Passhtrough: Disable, and no port forwarding
SIP Passhtrough: Enable, and no port forwarding
SIP Passhtrough: Enable+NET Helper, and no port forwarding
SIP Passhtrough: Disable, with port forwarding
SIP Passhtrough: Enable, with port forwarding
SIP Passhtrough: Enable+NET Helper, with port forwarding

None of them works for incoming calls.

 

[TOD]

Which port used to connect to your ITSP?
If it differs from the standard port(5060) or encrypted(over TLS), then all these tests were useless because the mechanism will not be able to identify such traffic and handle it.
Also you may try to enable Rport function on your IP-phone. Sometimes this can help ITSP's server to properly reply to client device.

 

[Nerre]

It's standard 5060 and no encryption.

I had it working years ago with my old router (an old PC running M0n0wall), with just port forwarding of the RTP port range.

What does Rport mean? The closest thing I find in the settings is Random ports...?

The phone is a Siemens Gigaset S675IP.

This is no high priority, since we have not had it working for 4 years. Outgoing calls work as they should.

 

[mmb]

I use it for VOIP.ms and it works great.

... and I don't use it with voip.ms and it works great : )

mike

 

[TOD]

What does Rport mean?

https://tools.ietf.org/html/rfc6314 Section 5.1.1
https://www.ietf.org/rfc/rfc3581.txt

The closest thing I find in the settings is Random ports...?

The phone is a Siemens Gigaset S675IP.

Perhaps this model does not support rport, or it is likely that supported and enabled by default(like in Gigaset C610A-IP), but there is no checkbox in the configuration.

Example of registration Gigaset C610A-IP(I have highlighted in bold, how rport support is represented in client-side and server-side):

Session Initiation Protocol (REGISTER)
Request-Line: REGISTER sip:xxx.xxx.xxx SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.xxx.xxx:5075;branch=z9hG4bK57d26187e910f32b289519d1bb6bdc0e;rport
From: <sip:xxxxxxxxx@xxx.xxx.xxx>;tag=2596272465
To: <sip:xxxxxxxxx@xxx.xxx.xxx>
Call-ID: 1965366488@192_168_xxx_xxx
CSeq: 382 REGISTER
Contact: <sip:xxxxxxxxx@192.168.xxx.xxx:5075>
Max-Forwards: 70
User-Agent: C610A IP/42.238.00.000.000
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Length: 0


Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Message Header
Via: SIP/2.0/UDP 192.168.xxx.xxx:5075;branch=z9hG4bK57d26187e910f32b289519d1bb6bdc0e;received=213.141.xxx.xxx;rport=5075
From: <sip:xxxxxxxxx@xxx.xxx.xxx>;tag=2596272465
To: <sip:xxxxxxxxx@xxx.xxx.xxx>;tag=98a7163a-ba340000-2cd57608
CSeq: 382 REGISTER
Call-ID: 1965366488@192_168_xxx_xxx
Server: Oktell 2.13.0 (HAL 202 May 5 2016)
WWW-Authenticate: Digest realm="xxx.xxx.xxx", nonce="66bb88e872313ab96334bcfd1b31a1e9", opaque="opaqueData", algorithm=MD5
Content-Length: 0


Session Initiation Protocol (REGISTER)
Request-Line: REGISTER sip:xxx.xxx.xxx SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.xxx.xxx:5075;branch=z9hG4bK195bf26c37581944e06b957c87c456a;rport
From: <sip:xxxxxxxxx@xxx.xxx.xxx>;tag=2596272465
To: <sip:xxxxxxxxx@xxx.xxx.xxx>
Call-ID: 1965366488@192_168_xxx_xxx
CSeq: 383 REGISTER
Contact: <sip:xxxxxxxxx@192.168.xxx.xxx:5075>
Authorization: Digest username="xxxxxxxxx", realm="xxx.xxx.xxx", algorithm=MD5, uri="sip:xxx.xxx.xxx", nonce="66bb88e872313ab96334bcfd1b31a1e9", opaque="opaqueData", response="9skw3u2p34fxe9nm2mdjgwedqx1yiw36"
Max-Forwards: 70
User-Agent: C610A IP/42.238.00.000.000
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Length: 0


Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 192.168.xxx.xxx:5075;branch=z9hG4bK195bf26c37581944e06b957c87c456a;received=213.141.xxx.xxx;rport=5075
From: <sip:xxxxxxxxx@xxx.xxx.xxx>;tag=2596272465
To: <sip:xxxxxxxxx@xxx.xxx.xxx>;tag=b8a7163a-1f350000-bc03bb0a
CSeq: 383 REGISTER
Call-ID: 1965366488@192_168_xxx_xxx
Contact: <sip:xxxxxxxxx@213.141.xxx.xxx:5075;transport=UDP>
Server: Oktell 2.13.0 (HAL 202 May 5 2016)
Expires: 3600
Content-Length: 0