Topic: IP Telephones behind a SME 7.0 Beta ?

[arne]

Hello !

I am doing some testing on using IP telephones behind a SME 7.0 B4 gateway.

I am using a Sipura and a Granstream adapter plus some (4) soft telephones for the testing.

I had a problem that the connection were lost for the two adapters after a few days of logon (2-3 days). I believe this was due to the fact that the Sipura and the Grandstream adapter used some of the same resources (UDP 5060).

I have now reconfigured the Grandstream adapter and forwarded some alternative ports via the SME 7.0. I believe (hope) that everything should work in a stabile way now.

The idea was to investigate how to set up one or more (10-12 ?) IP telephones behind a sme 7.0 Gateway.

If anybody has info or comments on this, please leave a message

Best reg Arne.

[torrestech]

Hi I use Cisco VOIP Adapter which has the Sipoura chipset. I found that it works fine except when making long calls. The adapter would restart itseld after about 15 minutes. It turns out that i had to forward the following ports to the adapter to overcome this problem.

TCP     5060     VOIP_ADAPTER     5060
UDP     5060     VOIP_ADAPTER     5060
UDP     53     VOIP_ADAPTER     53
UDP     69     VOIP_ADAPTER     69
UDP     20000     VOIP_ADAPTER     20000

Adam

[matsk]

Grandstream:
On www.voip-info.org is there a "Regular reboot" script for grandstream.
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone

Normaly you don't have to open ports to get sip phones to work and if sip phones fails to work behind a FW try to use a STUN server in the sip phone config first.

I have had ~15 IP phones/ATA's behind a SMEserver 6.5 without problems and without any port forwarded.


/Mats

[arne]

Thanks for interesting info. I have to admit that I even did/does not now what a stun server is. Found some interesing info here:
http://www.newport-networks.com/whitepapers/fwnatwpes3.html

[arne]

Some more interresting info:
http://www.voip-info.org/wiki-VOIP+PBX+and+Servers

[arne]

One of my ip telephone vendors recomend forwarding of these ports: UDP 123, 69, 5004, 3478, 5061 (Vendor name Televoip, Norway) I have experimentet a bit and found out it can work with forwarding of only port udp 5004 and udp 5060. Without one of these forwardings the comunication stops in a fraction of a second.

The other ip telephone use a tecnology that does not require port forwarding at all. (Telio, Norway).

I think that the technology behind these vendors are quite different and that Telio use some kind of external proxy, that Telovoip does not have or use the same way.

Of those two I like the Televoip the best because they leave all configuration and configuration access to the end user while the Telio solution is a "black box" with no user configuration access.

matsk -> Would you mind informing which vendor that you use for a 15 telephone line connection. I am 99 %sure that its possible to make such a connection with the televoip technology and proper forwarding to each telephone client. I have a 60 % believe that the telio solution can handle multible ip telephone connections without any forwarding. (Even though they dont inform about this. My believe is that such a solution is not proven and stable enough so they dont sell out such a product yet.

By the way I have not problem using my to ip adapters pluss the web based clients at one time.

[Franco]

Your answer lies in another question you posted about asterisk, with it you don't have to forward ports, let it handle the routing of your providers and only open the ports to a single system, be it SME itself or another box behind it. STUN servers are provided by your provider, and it would resolve the issue with NAT, which you wouldn't have in both setups above (unless your provider really requires it)

[arne]

One of my ip telephones use a external stun server and I have also tried to forward the same stun port for inbound connection as told by the sip telephone vendor.

According to info from the sip telephone vendor this will not free you from setting up forwarding on the gateway. According to their info it will still be required.

I have performed a series of experiments on this ip telephone, locking and opening one and one port with the external stun server access all the time and with and without the same port open for inbound trafic.

According to info from the sip telephone (line) vendor, for their system it will be needed use of stun server pluss five UDP ports forwarded.

According to my own experiments the connection works with use of stun server plus forwarding of two ports. Both those ports can be configured at the Grandstream adapter. Just now I am using UDP 5004 and 5061 (to avoid conflict with my other ip telephoes.

Because of these two incomming ports can be freely selected on the Grandstream adapter menu I belive it will be possible to install a number of ip telephones bihind one nat server (sme server) using this tecnology.

My other sip telephone vendor use a completely different technology. For their equipment it looks like there is only one port in use (UDP 5060) and this connection/adapter is able to work without portforwarding.

I think (guess) both use stun servers but they still works completely different.

One of the vendors (with the grandstream adapter) use en open produckt cansept where the user (I) can set op all parameters on the adapter. The other just delivers a "blak box" with no user access. (You just connect it and then it just works.)

I think there is quite different technology in use.

[arne]

By the way I think the ip trafic is rather different for the different tecnologies.

I have not studied it in detail yet, but my first impression is that some technologies use a simple trafic pattern where it is only a "simple stream" of "equal" udp packets wit the trafic set up from the inside sip client. I think there is also an automatic update routine programmed into the adapter that keep the connection to a external (sip proxy ?) server "fresh".

It is also my first impression that other tecnologies use another approach with a lot more complex trafic patern and different shear of tasks between clients and servers. (The clients will also have to do some server tasks due to a "missing" proxy or server function between clients.)

I believe that there is one type of sip connections that works more with a direct connection between clients while others are more dependent of some server functions between (and because of this the need for portforwardings will also be different.)

Have not realy done it yet but will try to dump and monitor some trafic to tro to observe some of those differences.

By the way, exuce me for my english. Some times even I can see it is not correct.

[arne]

If anybody is interested in looking into the trafic to from the iptelephones thhese tools can be used:

iptraf is intalled at the sme 7.0 by default.

ethereal can be installed via yum at the server-manager Works very nice and easy.

For reading the trafic to from a ip telephone at ip adress 10.0.0.7

tethereal -i eth0 host 10.0.0.7

Gives a detailed overview of how things are hapening.

Some other info/resources for further reference:
http://www.linuxhomenetworking.com/linux-hn/network-trouble.htm#_Toc92808515

[matsk]

matsk -> Would you mind informing which vendor that you use for a 15 telephone line connection. I am 99 %sure that its possible to make such a connection with the televoip technology and proper forwarding to each telephone client. I have a 60 % believe that the telio solution can handle multible ip telephone connections without any forwarding. (Even though they dont inform about this. My believe is that such a solution is not proven and stable enough so they dont sell out such a product yet.

By the way I have not problem using my to ip adapters pluss the web based clients at one time.

I'm using www.voop.no

And at home i have four different telephones (ATA's):
- Two connected to FreeWorldDialup
- One connected to a Swedish TISP
- One connected to a Norwegian TISP

And in addition to that I have three softphones (Pulver Communicator, X-lite and Firefly)

And no ports in the firewall is forwarded !

And some comments on the TISP's (Telephony Internet Service Provider) you mentioned:

Telio is unusable, you can't use them with asterisk or other hardware/software other than they provide, due to their propreritary implementation of sip.

And for televoip, I find it hard to understand why you MUST forward ports to get it to work with your sip implementation. But I have seen several scandinavian TISP's that has implemet their system in a way that causes problems for the endusers.

And there are a couple of other Scandinavian TISP's that doesn't work together with asterisk but that is a discussion we can take outside this list.


/Mats