SIP & Nat - one way sound problem

lev1987

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#1
Hi Everyone,
First of all thank you all for the great work and an amazing open source telephony solution, I'm quite new to Elastix and after setting up a pbx at home 2 months ago I've decided to try my luck with a bigger one at work.

I've set up 30 sip extensions and 30 iax2 extension, bought a E1 card to connect my pri, setup faxes & voicemails to mail, all works quite well and we've finally replaced our analog philips pbx.

The next step was to setup an external sip extension at home, Using a softphone (Zoiper), I've opened and fowarded all ports on my NAT (for my home static IP - will be reffered as: "199.199.150.234")

SIP: Softphone can register, when making a call the other side can't hear me, I can hear him perfectly.
IAX2: Works great no problem.

But I must use SIP because I want to connect an hardware phone and it can use SIP.

I know it's a pain in the ass and there are many threads about that but I could find nothing helpful (I've already setup sip_nat.conf)

I've attached my SIP debug logs when trying to make a call

External IP: 199.1.90.210
Internal IP: 192.168.100.254
MY softphone IP: 199.199.150.234

All help will be appreciated. http://forum.elastix.org/old_files/elastix_error.txt
 

dicko

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#2
Re:SIP & Nat - one way sound problem

The firewall(s), presumably the far end router and the host of the soft-phone must also pass transparently UDP/10000-20000 connections through from the Elastix Box to and from the soft-phone.

For confirmation that you have your SIP rewrites correct please post your sip_nat.conf
 

lev1987

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#3
Re:SIP & Nat - one way sound problem

dicko said:
The firewall(s), presumably the far end router and the host of the soft-phone must also pass transparently UDP/10000-20000 connections through from the Elastix Box to and from the soft-phone.

For confirmation that you have your SIP rewrites correct please post your sip_nat.conf
Thank you for the fast reply,
The firewall passes all ports to pbx (for the ip I'm testing from)

SIP_NAT.conf:

externip=190.1.90.210
localnet=192.168.100.254/255.255.255.0
 

dicko

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#4
Re:SIP & Nat - one way sound problem

Then check the firewall/router settings at the far end.

tcpdump is a good low level tool for debugging network traffic.
 

lev1987

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#5
Re:SIP & Nat - one way sound problem

how can I make sure asterisk actually uses the sip_nat.conf?
I suspect it might be ignoring it.
 

dicko

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#6
Re:SIP & Nat - one way sound problem

I believe it is a network configuration problemnot an Asterisk configuraton problem, so you need to debug at that level.

From your included sip packets and the via headers in them, in your debug file I don't believe it is a sip_nat problem.

As stated tcpdump is your friend, you will see the raw packets in and out, the address and port from and to also. Use it or something with the same functionality both ends, for example:

tcpdump host <far end apparent IP>
 

lev1987

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#7
Re:SIP & Nat - one way sound problem

dicko said:
I believe it is a network configuration problemnot an Asterisk configuraton problem, so you need to debug at that level.

From your included sip packets and the via headers in them, in your debug file I don't believe it is a sip_nat problem.

As stated tcpdump is your friend, you will see the raw packets in and out, the address and port from and to also. Use it or something with the same functionality both ends, for example:

tcpdump host <far end apparent IP>
I've used your advice, and apperently the port is not the same on both sides (screenshot attached), do you have any idea why?
the ip on the left is local, on the right it's my softphone.
 

dicko

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#8
Re:SIP & Nat - one way sound problem

That's normal, your problem is that there are packets only going one way, there are none coming from the far end. so somewhere a firewall/nat box is misconfigured.

I assume you see the bi-directional traffic at the start and the end of the call (the SIP traffic)
 

lev1987

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#9
Re:SIP & Nat - one way sound problem

I do not see any bi-directional trafic other then over 5060.

How come over IAX2 there is no problem at all?
 

dicko

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#10
Re:SIP & Nat - one way sound problem

because the signaling and the media load are all in the same connection port udp/4569 by default. only one connection to negotiate, it either works or not.
 

lev1987

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#11
Re:SIP & Nat - one way sound problem

do you know why there is not bi-directional trafic, the firewall is configured to Allow and Forward all to my asterisk box (only for my softphone ofc)
 

dicko

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#12
Re:SIP & Nat - one way sound problem

Unfortunately for you, that is a question for your maker (at least the maker of your firewalls) ;)

There are many ways of doing PNAT and unfortunately getting VOIP through some is harder than it should be, I suggest you try other hardware if possible.
 

lev1987

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#13
Re:SIP & Nat - one way sound problem

I use Checkpoint SAFE@OFFICE 500P, any known trouble with that?
 

dicko

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#14
Re:SIP & Nat - one way sound problem

Yep, they do SIP all wrong, later firmwares have apparently fixed it, as I say check with your maker.
 

lev1987

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#15
Re:SIP & Nat - one way sound problem

I have firmware 7.5.48x
any other ideas?
 

dicko

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#16
Re:SIP & Nat - one way sound problem

Personally I'd throw that POS away. I fought with them way to long to respect them any more.
 

lev1987

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#17
Re:SIP & Nat - one way sound problem

You really think it's all because of the firmware? is there a way around?
 

dicko

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#18
Re:SIP & Nat - one way sound problem

My way around was to replace one with a real firewall. I am obviously not authoritative and obviously biased, you will have to take it up with the arrogant bastards at Checkpoint.
 

lev1987

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#19
Re:SIP & Nat - one way sound problem

I hope there might be another solution :-\
By the way is there a symbian client that supports IAX2?
 

dicko

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#20
Re:SIP & Nat - one way sound problem

Don't know, I have no interest in that OS, but two seconds on google suggests fring.
 

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