Yate Community Forum
OpenLTE => YateBTS => Topic started by: ThePit on July 30, 2014, 02:18:12 PM
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Howdy,
I am running on Ubuntu 12.04 and with Yate 3.
I have set an Outbound connection through the NIB web interface.
I connect a GSM phone with a test sim made with pysim to YateBTS
Two test phone connected have no problem calling each other and having prolonged calls.
I can make outbound real world call.
Everything works for about 30-120 seconds.
After that, the real world phone can not longer hear the test Yate phone.
The test phone can still hear the real-world-phone with no problem.
A trace showed me that the problem appears to happen before my server that is the outbound connection.
I am currently connecting an Ettus USRP N210 directly to my laptop with Yate.
The laptop is then connected to the WiFi.
A TCP dump show that when hearing cuts out on the real-world-phone, packets from my outbound connection are still coming in; but packets are no longer being sent to the outbound connection server.
This makes sense given that I can still hear on the test phone but not on the real phone.
If I try to do a TCP dump on the Ethernet port, I cannot tell the difference between voice packets coming from the USRP and all the other packets coming from the USRP.
So, I am not sure if YateBTS stops sending the packets to the outbound or if the USRP stops sending them to Yatebts.
Is this just a limitation of the public code? Outbound calls will only work temporally for a proof of concept?
I have not tried to upgrade to Yate 4 yet since it just came out.
If anyone has any suggestion or comments, all help is welcomed.
Thank you very much.
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Try to update to latest yate.
Can you post a yate log?
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Howdy,
Attached is a log that I just created. I mildly edited it by replacing some of the numbers with #.
I erased my previous log.
Booted yatebts and connected a phone to it.
Made an outbound real world call and then I ended the log once the real-world-phone stopped receiving sound.
This particular test lasted longer than normal.
It took 2.5 minutes before the sound cut out.
I will have to upgrade to Yatebts 4 later as I must now temporary refocus my project.
I do plan to come back to YateBTS soon though.
I am mainly wondering if I am suppose to to be able to maintain prolonged real-world-phone-calls or if the public code has some thing built in to limit them.
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The log is useless. It is truncated after call start.
Please post a log with call termination signalling also.
Please start yate with timestamp logging (-Dt command line option would be ok).
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There is no limitation for call duration.
For the next log please start the message sniffer also.
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My bad.
Thank you for pointing out how to do better logs.
To start yatebts i used:
sudo yate -sd -vvvvv -Dt -l /var/log/yate.log
after starting I then ran:
telnet 127.0.0.1 5038
sniffer on
I connected a phone. Made a real-world-call.
After the sound cut out, I ended the call.
I waited a few seconds and then ended the log.
Thank you again for your time.
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Indeed, the log shows indeed sip reporting sent packets number to be less then received.
It might be a bug in mbts (or ybts).
We will investigate it.
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Can you test with new YateBTS 4 to see if it happens again?
Please post the log also.
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Howdy,
Currently I am making the rounds on similar BTS (such as OpenBTS) programs for comparison reasons.
I did save the state of my machine as it was with the logs that I already submitted.
I will go back to that state and update to YateBTS 4 to see if that solves the problem.
Unfortunately it will be about a week before I will get the chance to loop back to YateBTS.
YateBTS is definitely my front runner for BTS programs, but I still need to make some comparisons.
No matter what, I will reflash my machine to the saved state, upgrade to 4, and post the results.
I really appreciate all the help.
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Some more tests:
After loosing audio, can you make a new call without closing the other one?
What happens with audio in the new call?
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Took me a little longer than expected to get back to Yate.
I have upgraded to Yate 4 and all voice problem seem to be solved.
I was able to sustain a 30 min phone call to a real-world-phone without the audio ever cutting out.
Thank you again for your time.
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ThePit and we will continue to make it even more stable. It's a lot of work and we are rewriting some serious parts of mbts (the fork of OpenBTS we are using).
There are a lot of things going on to make this a carrier grade solution.
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Hello Thepit,
How are you? Can you share what gateway using for outgoing call and configuration.
Regards,
Tay
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Howdy,
Sorry, I cannot share my gateway.
It is a private server that I did not set up.
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Hello,
I had read that you can make outbound call successful.I would like to request that possible to post how you configure outbound section and nib? So, I can follow sample configuration.I hope to hear from you soon.
Best regards,
Tay
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Howdy,
I could post a screen shot if you wanted, but I would have to black out basically everything.
Sorry, I can't publicly share my server.
To make outbound calls though, once local calls worked, I only manipulated The "Outgoing" tab in the NIB.
I picked the protocol to be sip and I expanded it to the Advance version.
I filled in a Username/Password/Server/Outbound
My transport is UDP
I checked the boxes by Enabled/Match port/Match user
I hope that helps you.