News:

On Tuesday September 6th the forum will be down for maintenance from 9:30 PM to 11:59 PM PDT

Main Menu

voxbeam set up with OBI202

Started by rajp, December 28, 2013, 05:03:49 PM

Previous topic - Next topic

lk96

First of all it's not clear what you mean by "it's not working" in the scenario you mention.
The call is established but you can't hear anything? or one direction isn't working?
What happens if you dial into the AA from your obion app and you press the AA options?
does the tone go through?

Now some more comments:

1. I have given up on the Obion app some time ago for the simple reason that it seems
that the Obihai folks seem to have abandoned it and feels totally unsupported. There hasn't been
a release or an enhancement for a long long time. However, from what I recall there wasn't a
direct way to control the codec. Rather, you could specify the "quality" level and that
indirectly would give a hint to the app to use one of the supported codecs. So you have to
experiment and try a few of the settings and then monitor on your obi which codec turns out to be 729.
In case you care to explore more, I'm using the csipSimple VOIP client on my phone to dial
into the AA and then dial out. And obviously there are tons of other open sourced sip clients.

2. I think (but haven't tested) that you could possibly force the use of G729 by disabling all the codecs
except for g729 in the voice services -> obitalk -> codecprofile setting (you need to adjust the
corresponding codec profile to have only g729 enabled).

3. I don't think that "bridging" is your issue. Bridging here means that the two parts of the
call are "connected" at the obi. The way it works is that: you place a call into obi (that is one session)
and then the obi places another call to the remote end you want to dial (this is the 2nd session). When
these two sessions are established, Obi bridges them (which means connects them back to back)
and performs necessary transcoding if needed.
What I think is your problem is that the peer address of your Obion app shows up as 0.0.0.0
That makes me suspect that obi doesn't know how to reach the obion client and this most likely result
in a partially setup call. Now that's sometimes a bit hard to figure out.
In some cases I had seen this happening when the phone running obion
was behind the same home firewall as the obi.
As soon as I was switching to my 3G connection (which means that obion
was coming into obi from the outside and not discovered on the local network, things start working. So do make sure this is not the cause.

L.


rajp

My voxbeam set up is working fine but after 15 minutes my call drop from Voxbeam. I called there technical support guy and they told me that after 15 minutes server use reinvite and OBI device is not allowing to reinvite and after several try it drop my call. do you know where can I change configuration so that reinvite acknowledge call and it will continue my call?  Thanks for your help.
Raj

lk96

that's strange. I didn't have to do anything with re-invite settings. I have them
at whatever defaults and didn't experience the problem you see.

some of the SIP settings are under if you want to check them out.
ITSP Profile X -> SIP

I should point out two things though about my config with Voxbeam:
1. Voxbeam's SBC server doesn't support registration. So unmark the following;
Voice Services -> SPn service ->  X_registerEnable

2. Under the same voice services screen, I have enabled keepalives. Not sure if
it has anything to do with me seeing different behavior.


rajp

Thanks for your Reply.

I called Voxbeam and they reply me with following answer.

Per our phone discussion please see below the following trace. If you notice in the INVITE dialog there's an internal IP address that you setting in the Via header. When the re-INVITE gets sent back it's being pointed back to your internal IP address instead of your public. You need to either remove the internal IP address and replace it with your public address or completely remove your Via header from your INVITE dialog.

U 2014/03/02 16:06:30.802256 xx.xx.xx.x:5062 -> 108.59.2.133:5060

INVITE sip:0011101917600665927@sbc.voxbeam.com SIP/2.0.

Call-ID: 71456966@10.1.10.185.

Content-Length: 171.

CSeq: 8001 INVITE.

From: "Raj Patel" <sip:+14848759610@sbc.voxbeam.com>;tag=SP37f937057ffff0131.

Max-Forwards: 70.

To: <sip:0011101917600665927@sbc.voxbeam.com>.

Via: SIP/2.0/UDP 10.1.10.185:5062;branch=z9hG4bK-5cf5f497;rport.

User-Agent: OBIHAI/OBi202-3.0.1.4303.

Contact: "Raj Patel" <sip:sbc.voxbeam.com@10.1.10.185:5062>.

Expires: 60.

Supported: replaces.

Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE.

Content-Type: application/sdp.

How can I remove header Information? Anybody has any idea?

Thanks,
Raj