Far-End NAT UAs cannot complete calls to Near-End NAT UAs

[AirNavy]

1. Brekeke Product Name and version: Brekeke SIP Server 2.2.4.5/269

2. Java version:

3. OS type and the version: Win XP (Service Pack 2)

4. UA (phone), gateway or other hardware/software involved:
LAN (Near-end) : Cisco 7940 (SIP mode v8.9), X-Lite (SIP softphone), DI-624 Router + Cable modem
WAN (Far-end) : X-Lite users.

5. Structure
Server running on a PC : 192.168.0.100 (local) = 24.202.xxx.xxx (public)
DI-624 Router : 192.168.0.1 (local) = 24.202.xxx.xxx (public)
Cisco Phone 7940 : 192.168.0.110 (local)

6. Your problem:

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

Hello to all,

So I've been running this Brekeke SIP server for a few weeks, and quite frankly I think it does a great job. I managed to set it up quite easily opening and routing the right ports on the router, I was using X-Lite on both near and far ends. Boy you gotta love SIP/VoIP technology!

To add to the bliss, this weekend, I just clicked an ethernet wire between a Cisco 7940 and the router, configured it, (CallManager to SIP firmware conversion, TFPT... the whole thing!), everything works like a charm.

One minor thing now that bugs me. Probably a faulty NAT configuration. Since the beginning, I was able to initiate calls from the inside (near-end) to either near or far end UAs. Ring, answer, sound.
BUT... The opposite isn't true. Any far-end UAs cannot complete a call to a near-end UA, that is: within my LAN. Far-end UA initiates a call, rings on both sides but the minute I lift that receiver: not even a whiper comes through! The session timer starts on both ends, there is just no sound, neither on the callee or the caller's side.

If port routing was faulty, I assume that it would just not work at all.

Does this problem rings a bell to anyone?

Thank you,

Charlie.

[AirNavy]

** neither 'of' the callee 'nor' the caller....

Cheers!

Charlie

[lakeview]

Did you try RTP-relay?

Go to SIP Server's [Configuration ] -> [RTP] page.
[RTP relay] = "on".

Try it.

[AirNavy]

Thank you Lakeview, I'll definitely try this option as my config was showing:

[RTP relay] = "auto" ; and
[RTP relay (UA on this machine)] = "auto"

So anyway, I know put "on" and "auto" respectively.
I'll let you know if I get positive results.

While waiting for an answer on the forum I also happened to find out that my router's port forwarding/firewall required a bit more of fine tuning. Found out that I had to apply an "outbound" rule along with the "inbound" rule I had set up already.

I was still getting a bit of inconsistent results when receiving a phone call from outside my LAN. From a call to another (made in the same direction) it would still randomly work or not...

I am hoping for the best with this new setting.

Cheers!
Charlie

[lakeview]

Hi Charlie,

>> Server running on a PC : 192.168.0.100 (local) = 24.202.xxx.xxx (public)

Does it mean that the Brekeke SIP Server has two NICs for 192.168.0.100 and 24.202.xxx.xxx ?

Or is the Brekeke SIP Server behind a router?

>> that my router's port forwarding/firewall required a bit more of fine tuning. Found out that I had to apply an "outbound" rule along with the "inbound" rule I had set up already.

Is it the Near-end's router? or is it the Far-end's router?

[AirNavy]

Hello,


>>Does it mean that the Brekeke SIP Server has two NICs for
192.168.0.100 and 24.202.xxx.xxx ?

Let me explain my network infrastructure a bit. The SIP server does not have two NIC nor two IP adresses. The public IP allocated to my modem by my ISP is 24.202.xxx.xxx, (subject to change at any time if the lease time expires). Therefore all the world sees my array of components as one single IP : the 24.202.xxx.xxx.

All the other elements (including the SIP server) fall under my house
cable modem and router. The Local IP allocated by the DHCP-router to the SIP server is 192.168.0.100. In light of the above, that is why some port fowrading is required. As the SIP Server will only be visible on the WAN at 24.202.xx.xx I have to forward the right ports inside the router so they make their way to and from the SIP server.

(I had prepared a simple ASCII text-based schematic layout of my infrastructure to help you understand, however the forum does not seem to support this format as all the extra spaces are deleted. I could provide it to you by e-mail, should you need it)

>> Is it the Near-end's router? or is it the Far-end's router?

I am only involved in the near-end port forwarding/configuration. So I made sure the port forwarding occured on my side so that the SIP Server would be visible from outside my LAN.

On the Far-end user perspective, other than firewall management, would there be other considerations to such as port forwarding to make this work? For instance, I have configured a Cisco 7940 phone for an external use (i.e. outside of my LAN), gave it to someone so she takes it to her home, she connected it to her home router and I can see that the phone resgisters and TFTPs properly on the SIP Server. Piece of cake! Is there anything more to do on her side?


** New developments ** :

1. The system appears to be partially functional as some calls, regardless of the direction, will be initiated and sound will either flow or not. I must say that the "random aspect" of functionnality is only base on a per-call basis. Both phones ring every time a call is placed, so the INVITE process is not to be blamed. However, when you lift the receiver, it either works (sound flow) all the way through that call, OR one of the two parties cannot hear the other...

2. Independant from everything I have explained up to now, I also noticed that my aging 10/100 mpbs D-Link DI-624 router had a tendency to reset the 4 Ethernet ports all at once for a second when a call was initiated. After some reading on the subject I came to the conclusion that this was some sort of packet flood protection: an undesirable feature in a high volume packet flow system such as SIP/VoIP.

Unfortunately, the router I was using did not offer extensive configuration for packet flood protection. So guess what... Yes, I changed the router for a 10/100/1000 mbps router Linksys WRT310N. Since then, no more unexpected 1-second ethernet disconnections, but I have yet to see if this will solve other problems.

Cheers!
Charlie

[lakeview]

Hi Charlie,

>> I have to forward the right ports inside the router so they make their way to and from the SIP server.

The following port should be forwarded from the router to the SIP Server (192.168.0.100).
------------------------------------------
UDP:5060 (for SIP)
UDP:10000-10999 (for RTP)
------------------------------------------

If your router supports UPnP, you don't have to configure the port-forwarding.
Go to SIP Server's GUI: [Configuration] -> [System] -> [UPnP] and select [enable].
I know that Dlink and Linksys routers support UPnP. ( you need to use the latest router's firmware.)


>> On the Far-end user perspective, other than firewall management, would there be other considerations to such as port forwarding to make this work?

It depends on clients..

>> However, when you lift the receiver, it either works (sound flow) all the way through that call, OR one of the two parties cannot hear the other...

Try the RTP-Relay.
[Configuration] -> [RTP] -> [RTP exchanger] and select [RTP relay] is "on".


>> Yes, I changed the router for a 10/100/1000 mbps router Linksys WRT310N.

In my experience, Linksys's router is better than others for SIP/VoIP uses.