I have currently installed asterisk 14.2.1. I have one way audio for every codec . I have no NAT in sip.conf setting. can any one suggest me what should be the correct procedure to resolve it ?
Post some information about your network topology, with respect to the endpoint, along with your SIP settings, along with a capture of the SIP signaling showing the call setup.
On the sip.conf sample file there is a section called ;------------------ NAT SUPPORT ----------
give a read and you will have a better idea of what to change on your configuration file
http://svn.digium.com/svn/asterisk/trunk/configs/samples/sip.conf.sample
Try in the configuration file to register such parameter: nat=force_rport,comedia
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The reference to the configuration file was mainly about the external addresses, not the nat= parameter. For most common configurations, the default nat= should work.
In any case, the OP claims this is not a NAT case, in which case firewalls are the only real possibility.
Thank you very much everyone for reply.I will try suggested parameters and will ping if not work as per expectation.
Regards.
Sorry for late reply , actually I was working on different task.One way audio problem only occurs when it deals with “opus codec”. Another codecs which I use are ulaw , g722 etc. With those codecs call works correctly.Can anyone elaborate what actual problem with opus .As I mention earlier I am using asterisk 14 on ubuntu 14.04.