One way audio with Obi202, Panasonic KX-TGP550T04 and double NAT
Zopa:
Hello all,
Background:
I have Panasonic KX-TGP550T04 IP Phone which I've been using for years with CallWithUs.com as a SIP provider for outbound and Google Voice (via some sort of free DID redirect for GV) for inbound calls. This phone is on my NATed LAN. I have a static IP, although it's irrelevant.
Issue:
I have been working on configuring KX-TGP550T04 to use Obi202 as a SIP Gateway as per Option 1 in this manual http://www.obihai.com/docs/OBi-VoIP-Device-Attach-Legacy-IP-Phone-Workbook-v1-0.pdf, essentially putting my Panasonic IP phone behind a double NAT. SP1 is set to GV, SP2 and SP3 are not used, SP4 is set as SIP Proxy. I have this phone configured with a stun server of stun.callwithus.com: 3478.
Let's say I'm using 123456789 as a user name and my X_InboundCallRoute is set to {123456789>(xx.):sp}
My phone is configured with Registrar Server Address and Proxy Server Address pointing to 50xxxxx744.pnn.obihai.com:5063.
I'm able to call from my Panasonic KX-TGP550T04 to my Google Voice, but I get one way audio where I cannot hear the party on my Panasonic IP phone. The callwithus service is works just fine even with double NAT.
While one way audio is a common issue, I still can't figure out. Any advice is appreciated.
Additionally, after I get this to work, I would love to know what's the best way to configure additional GV accounts on SP1 and SP2 and allow seamless use of those if of GV lines on SP1 lines are already used up. If I have three GV accounts setup, does it mean I can have up to 6 concurrent outbound calls using GV?
TIA
azrobert:
I'm far from a network expert and don't understand your double NAT setup. Not that it matters.
I've been playing with a double NAT. I added a 2nd router with a different subnet to my network. I connected my Android to the secondary router via WiFi and registered CSipSimple to an OBi200 on the primary network. I was having audio problems.
I fixed the problem by doing this on the trunk CSipSimple was registered:
Service Providers -> ITSP Profile X -> General -> X_SymmetricRTPEnable: Checked
You might be able to get more than 2 concurrent calls with GV. I've only tried 2, but it's worth trying more. You will need to increase the MaxSessions. The default is 2. I think you should set it to the max number of concurrent users plus 1. This will allow for an inbound call when all users are on active calls. It can't hurt to make it more.
Voice Services -> SPx Service -> MaxSessions: 5
You can try a Trunk Groups to failover to another trunk. I only tested Trunk Groups for failover when a trunk is down, but it should also work when a call is rejected because of capacity.
Edit:
Maybe you should set MaxSession to 3, so the 4th call with failover. I not sure how this will work with Trunk Groups. You will just have to test it, unless someone else knows.
ianobi:
Within a Trunk Group if all the sessions for a trunk are in use, then the next call will failover to the next trunk.
However, in this case we do need some trial and error or expert help - SteveInWA? I have no idea how many sessions a GV trunk will support. In fact "sessions" may not be the correct description here as it refers to SIP. Setting MaxSessions to a particular number for an SPx Service supporting GV will be ignored as it is a SIP parameter that does not apply to GV.
Anyone using GV and Trunk Groups? Testers are needed :)
azrobert:
ianobi,
Thanks for the info.
Zopa,
I assume you will register additional IP phones on SP4.
You will have to increase MaxSessions on SP4.
Zopa:
Thank you for your replies!
azrobert, by double NAT I mean that I have my Panasonic phone connected to Obi202 LAN interface which is NATed to the upstream network (my local LAN), which in turn is NATed to the real Internet. So my phone has an ip of something like 192.168.100.101, Obi202 has an IP of 10.16.100.xxx and then my real internet address is something else entirely.
Let's keep the trunking of GV lines aside for a second, although I do agree it's a more interesting question to figure out. Even though I'm pretty fresh with Obi and advanced VoIP configs, I'm a very high level IT architect and would love to help with any testing, while learning things along.
However, right now I can't even get the basics of two way audio to work. I checked X_SymmetricRTPEnable first under ITSP Profile 1, then also under ITSP Profile 4, testing in between. Still have a one way audio, can't hear called party on my Panasonic phone.
What is the importance of this parameter? And which profile should it have been done under?
Any other suggestions?
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