Calls intermittently going directly to voicemail

[sr98user]

Russell,

Maybe I was not clear when I listed my NAT entries. My ATA uses 16384-16482 for both incoming and outgoing calls. It just increases by 4 after each call to get the next port number and cycles through the range again. I think this port number is communicated through the SIP packets. The adapter probably finds out the translated port number using STUN server if its used. I don't know a lot about how its communicated to the media gateway. The translated port number on the public side could be different depending on the router.

[VOIPoTim]

The translated port number on the public side could be different depending on the router.

The problem is that the majority of new routers don't handle the negotiation correctly and the ports are mixed up. So if the router is not expecting a connection on a given port, it will reject the incoming connection. Some routers do this properly, but it's rare to see it.

In theory users should not need ports forwarded, but again 99% of the time when we see the issues like this in logs, the logs for the user show that the packets were rejected by the router which in turn sent an error code back and it was passed upstream generating a drop to fast busy, disconnected, etc.

So you're correct that in theory, the routers should be able to negotiate. Most new ones rewrite all packets though and don't do it correctly so the ports aren't correct. Since this changes even with router firmwares (and we've learned most new routes auto update themselves), we can't really track which routers handle it properly and which don't (especially since we'd have to track every single firmware version too).

We don't provide networking support and recommend using connect our devices directly to their modems and bypass their consumer router and use the router built into the adapters. When they have issues that are 100% caused by a consumer router, we have to take the most direct approach to resolve the issues for our service. In this situation, it's to forward the entire range since that is effective 100% of the time.

Should users need to do it? No. As long as router manufacturers continue on the path they are on with adding all this unnecessary stuff (like SIP ALG, filtering every single packet and rewriting incorrect, etc), forwarding will eventually be needed by everyone or someone is going to have to invent workarounds.

Here is a perfect example where the stock Linksys router firmware for a brand new router completely breaks VoIP service and the user experienced all the standard router issues commonly caused by a consumer router (things like dropped calls, no audio, disconnected messages, missed calls, etc). Swapping to a firmware that handled it correctly and 100% of the issues were gone.

http://www.dslreports.com/forum/r221...Linksys-router

Router manufacturers are making things more and more difficult every day. Today I actually tell our team that the #1 thing that can make or break a user's VoIP experience is their router. It's no longer the internet connection because nearly all of those are fine for VoIP, but the majority of new routers require tweaking.

[voipinit]

Just discovered my routers UPnP feature when enabled opens the necessary ports. So for my router, as long as UPnP is enabled, I don't need to forward any ports.

[usa2k]

Granted this is related to OLD threads
http://www.dslreports.com/forum/remark,13561259
There have been modems that were the route cause of VoIP issues too.
(No pun intended!)

[Russell]

Russell,

Maybe I was not clear when I listed my NAT entries. My ATA uses 16384-16482 for both incoming and outgoing calls. It just increases by 4 after each call to get the next port number and cycles through the range again. I think this port number is communicated through the SIP packets. The adapter probably finds out the translated port number using STUN server if its used. I don't know a lot about how its communicated to the media gateway. The translated port number on the public side could be different depending on the router.

Yes, there are a lot of missing pieces of the puzzle which folks like me don't understand - e.g. what kind of negotiation of port number takes place. And, now that Tim has mentioned that "old" routers (which don't do any of the new fangled stuff work fine), how exactly do those routers know to let audio traffic from the media gateway in (e.g. is the ip address of the media gateway - different from the SIP server - part of the information that is exchanged in the negotiation)?

I have a really old router which worked fine with many VOIP companies and also with VOIPo until a couple of weeks ago. At that time I port forwarded. I'm tempted to undo port forwarding and restart all pieces after reading Tim's comment that old routers should work fine. Unfortunately, the wife volunteers three evenings this week and the phone needs to be "perfect" for any incoming calls - so it'll be next weekend before any such experiment.

My router does not appear to let me specify what type of traffic to forward - so presumably it's forwarding both TCP and UDP (and not just UDP).

[oldcqr]

I am getting very frustrated.

I find myself now needing to 'test' my service every few hours to ensure that it is working. Many times it is not. Today I find I am once again going directly to voicemail. Fri/Sat I was having problems with outgoing calls.

Most frustrating of all is that it appears that no one is even researching my problems. The ticket over the weekend was replied to with the 'forward all the ports' reply. I've had the port forwarding in place for several weeks now.

I know the Voipo staff can tell if the ports are forwarded or not, since they caught that I had not forwarded them on a previous ticket (I had the entry but forgot to enable it). That tells me the rep didn't even bother to check anything.

[VOIPoTim]

I know the Voipo staff can tell if the ports are forwarded or not, since they caught that I had not forwarded them on a previous ticket (I had the entry but forgot to enable it). That tells me the rep didn't even bother to check anything.

Unfortunately, they can only tell if the SIP port is forwarded. There is no way for us to know if you forwarded all.

If it has ALG or a SPI firewall, these will also likely need to be disabled.

I know it's frustrating, but unless the router allows the traffic through fully, there's not much we can do on our end. We can't really force your router let traffic through from our end. We're happy to help you review the router configuration to see what can be adjusted to open it up.

If you have a ticket number, I'm happy have a Tier II rep contact you to review your router settings.

[oldcqr]

I just opened up remote administration to my router, and placed the information into the ticket I have open for your support to access it.

PPV-307843