Don't hear ringing of called phone

[SpaethCo]

I'm not sure how much detail you really care about here, but this can actually be tracked back to one of a few different things.

Not all audio you hear comes from the network
For example, when you pick up the phone you hear a dial tone, but that's actually generated by the device that VOIPo sent you. In fact, that device can generate every sound the phone network uses from ringing sounds, to busy signals, to the sound you hear for call waiting.

Not all calls are handled the same
This is the part that gets confusing. The ringing that you hear on a call can actually be generated in 2 different ways. Say you want to call 444-555-1234. After you finish dialing the number on your phone, the ATA device generates a request to the VOIPo servers that looks a little like this:

INVITE sip:14445551234@someregion.voipwelcome.com SIP/2.0
{A bunch of other stuff goes in the packet guts, mostly things like your username, what codecs your device supports, and what UDP port you will receive audio on}

VOIPo's servers can respond back with something like what you see below. The "180 Ringing" response means that your adapter should generate the ringing sound.

SIP/2.0 100 Trying
SIP/2.0 180 Ringing
SIP/2.0 200 OK
{ This signals that the call has been answered and gives the remote IP handling the audio for the call }

... but they can also respond back with something that looks like this:

SIP/2.0 100 Trying
SIP/2.0 183 Session Progress
{ This is a bit unique in that you get an IP for call audio before the call is answered }
SIP/2.0 200 OK
{ Here the call is answered, and you get an IP for where the audio will go to/from -- it's usually the same as the Session Progress IP }

This option uses what is known as "early media" because you get audio from the PSTN before the call is answered, so in this case the ringing sound actually comes directly from the PSTN. This is essential for dialing things like AT&T 800-number teleconference services because AT&T doesn't actually signal the call as being answered until after you've entered a valid conference pin code.

As was already mentioned, a lot of routers include a feature called SIP ALG. What it's supposed to do is read all of the SIP status messages and read things like what port your ATA says it will use for a call, and what IP your provider says will be used for the call audio, and automatically open a hole in the firewall and create a NAT mapping to get that traffic to the ATA. The problem is that SIP ALG sometimes has problems reading the SIP packets, and sometimes skips some altogether like the "183 Session Progress" packets -- when that happens, you have problems with incoming audio.

Since you're having intermittent problems, as an outside observer with limited knowledge of how VOIPo is setup these would be my first 3 guesses:

1) Your adapter isn't properly generating a ringtone for calls made on carriers that don't support early audio. If you go to your Call History and click on calls you can see a reference ID for each call. If you can provide to VOIPo support which calls had ringing and which calls didn't, they should be able to track down the carrier they used for each call and use that to troubleshoot further.

2) You have a router in front of your ATA with SIP ALG or similar functionality and it's eating some of the "183 Session Progress" messages, or preventing the correct NAT mappings from being setup.

3) VOIPo has a rarely used carrier that is being used to terminate some of your calls, and VOIPo's SIP gateways aren't properly translating call progress codes when relaying status to your ATA.

[christcorp]

Spaeth; excellent comments. There's a lot about voip that people don't understand, and when there's a problem, it can be difficult to discern where that problem is. I noticed when VoipO was first starting up and I was Alpha testing with them, we had a certain problem. After a lot of experimenting, I found that the problem only existed when I called a certain state. The problem got narrowed down to one of the partner providers passing the packets. So that too is a good experiment to try. Do you get the no ringing with ALL calls, or only certain people/places you call. Not saying that a customer should have to know how to do any of this. They shouldn't. But voip is still a novelty/niche market. It's not something my 80 year old parents are going to buy. Matter of fact; probably 98% of the 2000 people who work with/for me have traditional phone service. Of the other 2% there's those with "Digital Voice". (Cable company's version of voip, but only in town. They hand off to the Ma'Bell). Besides me, maybe 2-3 others have "True Voip".

Another part of Voip that most people don't understand is that to make a voip phone call, it takes 2 separate parts. You have the SIP (Session Initiation protocol) Which is the part that initiates the call. It usually goes through the voip provider. (I am not 100% if VoipO is doing it this way, but most do). You connect to VoipO via the internet, they send your request through their network of leased lines to the destination you are trying to call. The SIP initiates the handshake. Your request is usually then handed through Ma'Bell (PTSN) because most calls are to a land line or cellular. Once the initiation has been established, the 2nd part kicks in. That's the RTP. (Real-time Transport Protocol). That's the part where your actual voice packets in the conversation is attached to. The UDP protocol. That traffic however doesn't follow the SIP. It normally doesn't go from you to VoipO and back out. Once the SIP has the initiation established, the RTP takes a direct route (As direct as the internet can be considering they use LCR for traffic) to the handoff gateway at the PTSN. Then your voice is converted back to analog and on to the final phone.

The problem of not hearing the other end ring in your phone is not uncommon. Unfortunately, the cause is also uncommon. You'll find that some are in the Router, having SIP ALG and SPI turned on. Some routers it's a problem; some it isn't. Some, the voip adapter actually needed a firmware upgrade. Some had port problems with the customer and the server. (They changed the server in the voip adapter config and it fixed it). Some had problems in the proxy settings using a domain name and it got corrected by putting in the actual IP address. (DNS Issue). Bottom line, there can be numerous reasons for this problem. But if it's any consolation, a simple google search will show that this problem has been reported by customers of just about every voip provider on the planet. It sucks when it happens to us personally; but the problem isn't personal.