Half-Duplex incoming calls (outgoing is fine)

[wr5678]

Hi,

I've been using an OBI-202 for a while with no problems.  Had to switch ISPs, gotten a new router, and now I've got some odd behavior.  I haven't changed SIP providers, just the router/isp.

When I place a call everything works fine, I can talk with the called party no problem.  Incoming calls ring, I can answer fine, however I cannot hear the caller (but they can hear me).

It's pretty much a generic setup for my SIP provider (callwithus) as far as the Obihai device goes.

In trying to troubleshoot this I've tried port forwarding the whole range of RTP ports (16600-16798) in use for the service provider to the OBI device, that didn't seem to help.  No problems in SIP registration, no ALG on the router.  (FRITZ!box 7360)

One thing that seems kind of interesting is that it appears that the ISP is running ipv6 and supporting ipv4 clients by tunneling them through it.

Any other thoughts/ideas on other things to try?

Thanks

[drgeoff]

Does callwithus have a test number or echo test or voicemail or any other service that sends audio to you?  If yes and you call that do you hear the audio?

[wr5678]

Yes, when I place a call (in this case to a person, not a test number) it works fine.  If the same person dials me it doesnt....

[QBZappy]

@wr5678

You might consider connecting the OBi directly to the modem to see if it is router or ISP related issues.

[sdb-]

Do you get a publicly routable IPv4 address?

If they really are tunneling IPv4 inside IPv6, and/or if you do not get a publicly routable IPv4, that means you are behind carrier grade NAT and it may be a challenge to get SIP working.

For any NAT related issues, forwarding ports is only one knob, and it usually will only fix things if the router is losing the connection state.  Putting the OBi directly to the modem is the best test for this issue, and forwarding ports is what we use to fix it for day-to-day use if eliminating the router fixes the problem.  If cannot eliminate the router, disabling all port forwarding and setting up the OBi as the DMZ host is the next, best test.

Configuring a STUN server is another knob. It might help here.

The router may have a SIP ALG (application gateway) which it uses to be smart about SIP.  Most of the time it is better to disable the ALG and let the SIP endpoints use their own intelligence.  But whichever way it is set now, try the other way.

Oh, and best to only make one change at a time...  E.g. DMZ and changing SIP ALG might leave it broken when one or the other would have fixed it.

[wr5678]

The "router" in this case is a DSL modem, it's own SIP device (providing a "landline" from the ISP), and a WiFi AP all-in-one, so plugging directly into the modem is a non-starter.

The device does get a routeable IP address, I've tried making it a DMZ host (there is an option to forward incoming to a single host on the network in the router), that didn't make a difference.

If it's some sort of routing problem, why do both RTP streams come up and work for an outgoing call, but not for an incoming call (i.e. the caller hears me, but I don't hear them).  I have a SIP trace of an incoming call in this state if someone from obihai support wants to look at it.

[lhm.]

Try this,

Switching your OBi's Ethernet ports to 100 mb/s full duplex:

1. Dial *** (wait for response) then press 0
2. Enter option 27 and press #
3. Press 1 to set a new value
4. Enter a value of 1 and press #
5. Press 1 to confirm/save
6. Hang up
7. Wait for the OBi to reboot

[wr5678]

I doubt that the duplex on the ethernet side is the issue, but I tried the 100mb/full duplex thing anyway.  Didn't make any difference.

Neither did setting up a STUN server.

[giqcass]

The router has its own voip system in place.  I wonder if that somehow causes the problem.  Perhaps switching to non standard ports might help. Instead of 5060 maybe 11116.

[wr5678]

Contacted my SIP provider and tried an alternate port, also switched from UDP to TCP.  Neither helped

[sdb-]

The device does get a routeable IP address, I've tried making it a DMZ host (there is an option to forward incoming to a single host on the network in the router), that didn't make a difference.

That does not make sense.

What device gets a routeable IPv4?

What device did you make a DMZ host?

If it's some sort of routing problem, why do both RTP streams come up and work for an outgoing call, but not for an incoming call (i.e. the caller hears me, but I don't hear them).

Not specifically a routing problem, but because of NAT.  Any NAT can mess up SIP leading to what you describe.  Routeable address is just a way of detecting NAT.

Any addresses in these ranges are not routeable across the Internet: 192.168.x.y/16, 172.16-31.x.y/12, 10.x.y.z/8.  Technically anything can be made not-routeable by your ISP, and there are a few others that sometimes get used, but hopefully not often as services using those addresses will be broken...

If you visit a "what is my ip" page, the displayed IP address for you should be the same as shown on your PC network interface properties if you have a routeable address to your PC, or your modem/router's WAN address if it is the routeable address, or something totally unlike anything on your devices if carrier NAT or going thru a web proxy.

If you have a routeable address on every device it means you don't have any NAT.  If you have a routeable address only on your modem/router's WAN interface then the NAT is done by your modem/router and hopefully under your control. If nothing you have gets a routeable address then you have carrier NAT and all bets are off.

[wr5678]

The router WAN interface has a routable IP address.  I have the WAN interface on the obihai set for DHCP, and it gets a private ip address from the router which is performing NAT.

In the port forwarding rules for the router no combination of ports i've found (up to and including putting the obihai into the DMZ) makes the issue I'm having go away.

I can always make outgoing calls, but I never get audio on my side of incoming calls, the caller can hear me.

Tried alternate ports with SIP provider, STUN, no STUN, looking for other ideas.

[MikeHObi]

Did running the VOIP test give you any indication of an issue?
http://myspeed.visualware.com/indexvoip.php

[Mango]

Incoming calls ring, I can answer fine, however I cannot hear the caller (but they can hear me).

Half-duplex audio is where the other party is temporarily muted while you're speaking, and it's impossible for them to interrupt you - a la walkie-talkies.  What you're describing is one-way audio, which may be why you've received some suggestions that didn't work.

May I suggest you set the following:

Service Providers >> ITSP Profile x >> SIP >> X_DiscoverPublicAddress: (unchecked)

Also, disable STUN.

My guess is that CallWithUs is trying to negotiate direct media, which is failing.  I hope this will force it to proxy the audio.  Let us know if this solves the problem or changes the symptoms.