I have an Asterisk installation.
All extensions work well, both local and remote, if bandwidth is used.
On mobile networks (3G, 4G), there is no audio after connection.
In the CLI it is shown:
“switching from simple_bridge technology to native_rtp issue”
Only when using mobile network.
Can anyone give any tips?
I’ve been researching a lot, for days and nothing!
I have other servers running, with the same settings (copy) and they work perfectly.
Thanks in advance for your help!
Not without more details of your configuration. For that to happen, it sounds like the mobile network is connected using the same technology as the caller, but other destinations are not. We cannot even tell whether that technology is DAHDI, SIP, or PJSIP.
I get two handsets, both with the same SIP PHONE, as Linphone, Zoiper, etc.
If both are connected to a broadband network, it works 100%.
If on those same handsets, disconnect from the broadband, connect to a 3G network and make a call, they are mute after connecting.
Here are the settings (sip.conf):
canreinvite = no
udpbindaddr=0 0 0 0
pickupgroup = 1
pickupgroup = 1
You don’t have either PSTN or Mobile network entries in the configuration you have given!
I do not use PSTN.
The asterisk is accessible by a public domain, with ports 5060, 10000-20000 (rtp) open and tested.
Actually, I have two other identical installations, running on other servers / domains but they do not have this problem.
I do not understand why the fault occurs only with 3G type connections.
So you are not trying to call numbers on mobile networks, but simply using them for data?
That sounds like it could be deliberate blocking of cheap calls.
Have you ruled out bandwidth limitation and delay on the 3G data network?
If you are not using a high compression codec like g729 the demand for continuous bandwidth and maximum delay between packets may exceed some 3G networks available here where I live.
Sorry, I guess I could not express myself accurately.
I refer to mobile networks as a means of connecting to the server (internet).
Communication is always SIP-asterisk-SIP, thru asterisk server.
In the actual example I mentioned, the 2 handsets have an internet connection and dial SIP between them via my asterisk server. It happens the ring, answer, but there is no audio between them.
Thank you jcpierri.
Do you think that low bandwidth would cause total silence?
Well, I can make calls with success to my other servers with to the same 3G network, but I’ll try to use g729.
You need to get the protocol trace and check the addresses in the SDP. You also need to use rtp debugging to see if anything is being sent or received.
Thanks - I’ll check it out
Low bandwidth can cause audio delay, and distortion, but total silence seems to be no audio flow at all, it is good you start doing and rtp debug as was suggested on a previous post
Ok - thank you.
I’m going that way
the problem is with your ICP its blocking RTP packets and your device is behind NAT u need to check any firewall before and after the asterisk server and check your ICP policy of VoIP service.