QUESTION Vitelity TCP

[billsimon]

Does anyone know how to configure a Vitelity trunk (on the Vitelity side) for TCP?

I can go to a DID and change routing to SIP over TCP. At the end, Vitelity gives me an IP address and states that calls will come from that IP over TCP.

The IP provided does not allow registration, even though that is the only method I have set up for this account.

I know a lot of folks here use Vitelity. Has anyone tried TCP? Is it only available when receiving calls at a static IP (i.e. no registration)?

 

[GlenC]

I tried setting that up today and I was never successful. Doesn't seem to be much helpful information anywhere on the net. Vitelity's support documents seem to center around username / password auth. Were you ever able to get it setup?

 

[GlenC]

Well, in case anyone is following this, I got this reply on a ticket submitted.

Hello,
The TCP routing only works for Microsoft Lync. You will need to use UDP for any other device.
Please let us know if you have any other questions.

Edit: and then I received this response, from another tech, I still can't get it to work though:

Hello,

That's not entirely true, if you set tcpenable=yes in your /etc/asterisk/sip_custom.conf it will work for Asterisk, but we are unable to support it if you run into issues. I did just get it working on my own device last week

 

[billsimon]

No, I didn't get it working either. I have Asterisk configured to use TCP with tcpenable=yes (in Asterisk SIP Settings). But it's still not clear whether their TCP service allows registration. My TCP traffic was rejected. The tech calls it "TCP routing" which implies (in my mind) SIP TCP to a SIP URI, or static IP address -- not registration. If you get any more feedback, let me know. For now I have given up on it.

 

[GlenC]

I did get another response back from him. I probably won't have time to mess with this again until next week but this is his response when I asked him what to do with the IP I was provided when I switched on TCP there. Before you try this, you might want to get IP authentication working with UDP first and then switch it over to TCP. At least then, that part you will be sure of. Here's his response:

It would be best to open ports 5060 and 10k-25k to that IP, or whitelist it in any firewall settings. In my tests I didn\'t need the tcpbindaddress or transport settings just the _custom file settings. You can also run a tcpdump capture (tcpdump -s 3000 -w vitelity.pcap port 5060 or portrange 10000-35000 ) on your end point to see exactly what traffic is reaching it, which is how I identified the invites were indeed hitting my pbx.

 

[billsimon]

I already have it working fine with UDP. That last message you got tells me they are doing static forwarding (SIP URI) and not registration. Otherwise there is no need for port forwarding, firewall settings, etc. Not what I am looking for, so I will stick with the UDP setup.

My goal is to configure my PBX with TCP SIP only, which is no problem for my phones and softphones (extensions), but surprisingly unsupported by most of my providers.

 

[GlenC]

I started having second thoughts about it today. I had it port forwarded today while I was messing with it and my linux firewall was getting hammered on port 5060. I'm not so sure the reward is worth the worry, if there's any reward at all.