Hi,
I have a one way audio issue if i use my test dialer (sip client) on a mobile data network. But, if i use the same dialer when on my Wifi network, audio is fine. What could be it? I am running asterisk 16.30.1 ; So all my production calls are dropping after “rtptimeout” value set on sip.conf.
I have my rtp.conf with the below details as previously suggested.
rtpstart=10000
rtpend=60000
icesupport=true
stunaddr:19302=stun.l.google.com
What I understand is you are using two soft phone. If you are in you LAN you can communicate to each other and if you are connected using mobile network the voice is one side.
While you are on mobile network the traffic is flowing through NAT where as in your LAN it is not. Please check your NAT settings and also check if the packets are getting blocked any where.
Caller is a softphone (connected to mobile data or wifi) registered with a voip switch. Called party is a real gsm number. I am calling via a sip provider. My call flow is like below:
Caller (softphone) → Voip switch → ISP router → Asterisk —> Sip provider
When my caller softphone is on wifi, audio is just fine 2 ways. But, when i am using my data to connect my caller softphone to voip switch and make the same call (as like when on wifi), i can hear the B-side audio. But he cannot hear me (the caller).
Any suggestions?
Hi,
It should be a problem with the network. Can you capture the log from the soft phone in both the scenario and share please.
Regards,
Partha
It works now. I had “media proxy=auto” set on my voip switch. I changed it to “media proxy=on” and now audio is fine on both mobile data and wifi.
Thank you for the time. Appreciate it.
Great… enjoy.
Regards,
Parth
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