Linksys PAP2T voip Ports

[clarkovsaturn]

I just got a list of voip ports from Linksys support:

Haidee (22996): These are the ports that the adapter uses for VOIP:
5060 to 5061 UDP
53 to 53 UDP
69 to 69 UDP
10000 to 20000 UDP


I gotta say, I'm impressed with Linksys for having immediate text chat live support. The wait was very short, and they email you a chat transcript afterwards.

I got these after calling Belkin support about getting the PAP2T to work properly with the Pre-N router (I'm still testing to see if I have to send the thing back or not!).

Belkin told me if I open those ports it should work fine.
Viatalk told me to shut off the firewall on the Pre-N, and to put the PAP2T in the DMZ.

Hopefully I'll eventually get to the bottom of this. The big question: with the Belkin Pre-N be able to work consistently? Right now I have it working... we'll see tomorrow!

[DracoFelis]

I just got a list of voip ports from Linksys support:

Haidee (22996): These are the ports that the adapter uses for VOIP:
5060 to 5061 UDP
These are the default SIP (call setup) ports.

I personally have my router forward the SIP ports to my VoIP adapter (in my case an SPA-3000, instead of the VT supplied PAP2). Ever since I've done so, inbound VoIP calling has become more reliable. Which makes sense when you think about it, because inbound calls always send a SIP packet to you, in order to tell your adapter to ring your phone. And if your router is blocking that SIP packet for any reason, your phone can't ring as a result!

53 to 53 UDP
Isn't that the DNS port?

About the only connection I can see with VoIP, is that (like all other internet clients) the adapter has to do a DNS lookup to resolve the IP address of any internet (name) addresses it is trying to reach. However, since this is an outbound request only, most routers will open up the port for the outbound DNS connection.

As a result, I do NOT forward this port in my router.

69 to 69 UDP
This is the TFTP port. It's only needed if/when you are having ViaTalk "provision" your adapter. If/when this port is blocked, the adapter itself will still work (including being able to make/receive calls), but the remote "provisioning" feature won't work as expected.


10000 to 20000 UDP
Are you sure that 10000 to 20000 is correct? On my SPA-3000, the default RTP (voice) ports aren't 10000-20000, but are instead 16384-16486 (although the VoIP adapter lets you change them if you wish). I suppose LinkSys might have used different ports for the PAP2, but it seems a little strange to me.

BTW: Do you have the admin password for your adapter? If so, you should be able to tell which UDP ports are used for RTP by looking at the values of the "RTP Port Min" and "RTP Port Max" on the SIP tab of your adapter. Those two values should control the range of RTP ports used for the voice traffic.

FYI: In addition to forwarding the SIP ports, I also have my router forward the RTP ports being used to my VoIP adapter. The theory here, is that forwarding those ports doesn't hurt, and might help some to avoid issues with the voice stream.

[Agrajag]

ViaTalk often sets the RTP ports to 10000-20000. In my various setups with them, I've been all over the RTP spectrum. Right now they have me at 18384-18486 and 19384-19486 for my two lines.

[connervt]

Yes, ViaTalk has the ability to move RTP Port Min and RTP Port Max around, if they are remote provisioning their supplied ATA. In this case, you likely need to open 10000-20000.

If you are BYOB, you could easily go with the Sipura defaults of 16384-16482.

[DracoFelis]

Yes, ViaTalk has the ability to move RTP Port Min and RTP Port Max around, if they are remote provisioning their supplied ATA. In this case, you likely need to open 10000-20000.

If you are BYOB, you could easily go with the Sipura defaults of 16384-16482.
Good to know.

Of course, the key point here is that the ports aren't 10000-20000 (or 16385-16486) per se. Those ports are whatever port range you have chosen in your adapter (on the "SIP" tab). If VT defaults this to 10000-20000 (which is also the http://www.asterisk.org default, if I'm not mistaking), than that is important to know if/when you are using a VT provisioned adapter. However, if you are doing your own setup (either BYOD, or VT's adapter with provisioning turned off), than you have a LOT more flexibility in which RTP (voice path) ports to use.

NOTE:
You do NOT need 10000 ports open for RTP! RTP only uses 1 port for each open voice channel, and the adapter can handle at most 4 voice channels (each of the 2 lines involved in a "3-way call") at a time. So even if you leave a lot of "room to spare" for some ports that are in use elsewhere in your LAN, you can still get away with restricting your adapter to a range of maybe 20 UDP ports (which is a far cry from the 10000 ports that are used "by default"). And IMHO restricting the port range is in fact a very good idea, if you are going to "port forward" to the VoIP adapter (as restricting the port range on the adapter, means that your router has to "forward" a much smaller range of UDP ports to your adapter)!