One way audio with Obi202, Panasonic KX-TGP550T04 and double NAT
azrobert:
Repeat. I'm not a network expert, so anything I say might not be accurate. If anyone else wants to chime in, please do.
Look at the OBi Call Status when a call is in session.
Status -> Call Status
You should see RTP packet counts like this: tx=213; rx=222
When you have one way audio, one of the counts usually will be blank.
This is usually a port forwarding problem.
In a single network you should port forward the RTP ports to the OBi's IP address.
It's been awhile since I played with double NATs, but I think you need to port forward the RTP ports in both networks.
The primary router should assign a 2nd IP address for the OBi.
In my case the RTP ports were flowing in both directions.
I decided to change any parm related to RTP one at a time.
I don't know what's Symmetric RTP, but it was the first parm I changed and it fixed my audio problem.
SteveInWA:
Just commenting on the GV aspect as it relates to OBi use:
I think there are too many "balls in the air" to troubleshoot this -- I'd start by diagnosing a plan, vanilla configuration of the OBi plugged into the primary internet router, with an analog telephone plugged into the phone port, and testing if two-way audio works on both outbound calls (which are using Chat/XMPP directly with GV's infrastructure), and with inbound calls, which you are forwarding to as SIP DID. If both work fine, then move onto the next stage of complexity. If inbound calls work, but outbound calls have one way audio, then we can troubleshoot that, for example.
One reason I suggest this, is that there have been some recent (in the past week) issues with the audio stream dropping on some users' GV calls. Google Engineering is investigating that issue, which could be a red herring, confusing your troubleshooting.
I also don't profess to be the VoIP network architect/expert, so I will leave that conversation to y'all. I will just remind you that GV doesn't use SIP to set up or control call signaling, so don't bother futzing with SIP settings for GV SPs. I don't know how the old Chat/XMPP service works or doesn't work with ICE, STUN, NAT or other network traversal techniques, sorry. One other thing, don't forget to disable SIP ALG in your router, if your calls via your SIP DID are the only ones having one-way audio.
Regarding multiple calls using GV: You can only register and use one GV account per OBi SP, so on the OBi 202, that gives you support for a maximum of four GV accounts/numbers. GV supports call waiting when forwarding calls to a conventional telephone carrier or SIP DID, as long as that carrier supports call waiting. I don't know the maximum number of calls that can be juggled this way; it's not documented.
Assuming that you have multiple GV accounts, and you want to use them with your OBi, you would log into the OBiTALK portal page, with your OBi device added, while also being signed into the desired GV account on the same browser session. You'd then simply repeat the GV setup procedure for each OBi SP you want to assign to a GV account. The procedure will even display the GV account it's about to connect, to make sure you are selecting the right one (it's easy to accidentally select the wrong account, since Google supports multiple accounts being signed-in on one browser session).
You probably already figured it out, but just in case, here are my instructions for setting up GV:
http://www.obitalk.com/forum/index.php?topic=8560.msg56460#msg56460
Zopa:
Was just about to try heavier troubleshooting and ... everything is working. Must've been GV... Thank you for assistance. I thought I checked everything, but still.
Now on to trunking GV accounts. :)
azrobert:
Voice Services -> Gateways and Trunk Groups -> Trunk Group 1
TrunkList: sp1,sp2,sp3
DigitMap: (Msp1)
X_InboundCallRoute: {123456789>(Mtg1):tg1}
Define ITSP A, B and C DigitMap with the same rules.
It has to be setup this way.
Trunk Groups work differently.
ITSP A DigitMap must match the dialed number to route a call out SP1
Same for ITSP B and C
If you are calling from the Phone Port change the Primary Line to TG1.
I have GV defined on SP1 and set MaxSessions=1
The second call failed over to SP2, so MaxSessions does apply to GV.
Zopa:
Thank you.
I think I like the MaxSessions=1 on SP1, to balance CallerID vs. keeping a channel open for incoming calls. Essentially because of GV's 2 channels (in any incoming/outgoing combo) limitation, and my desire to have a specific number to show up on caller ID most of the time, I have to compromise. Therefore I would want the first outgoing call to go from GV account 1, while all other outgoing calls going to GV accounts 2 and 3. This way the second channel of GV account 1 will be open for an inbound call.
From what I understand while free, GV is limited in a number of way, such there is not way to have an arbitrary CallerID passed and not failover once the limit of two calls reached.
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