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I am trying to host a conference meeting using Asterisk's ConfBridge. How to handle nat for a confBridge application like it is handled in sip.conf by specifying 

nat=yes

Is there a way to configure something similar for confBridge. I went through confBridge.conf comments. But couldnt find any help.

The issue I am facing is that I am receving only one directional audio when I try to join 2 participants. What could be the possible reason?

EDIT:

Configuaration of the sip peers from sip.conf I have the sip peers configured correctly I guess. Here is my sip peers configuration.

[5001]
type=friend
username=aki
secret=password
callerid=aki<5001>
host=dynamic
context=t***e
nat=yes

[5002]
type=friend
username=apu
secret=password
callerid=apu<5002>
host=dynamic
context=t***e
nat=yes

This might help to get a better perspective.

I am writing my own sip client using jain-sip. The same configuration works well(bi-directional audio) when my sipclient and a 3rd party sip softphone are communicating directly through Asterisk. It is only when I use ConfBridge that the audio from a 3rd party sip softphone to my sipclient is not audible. I have tried different sip softphones and still the result is the same.

Akshay Kasar
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1 Answers1

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You not need deal with nat in conference.

Conference works with upper level channels(sip/iax). So you need configure correctly sip or iax peers

arheops
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  • I have the sip peers configured correctly I guess. Here is my sip peers configuration. `[5001] type=friend username=aki secret=password callerid=aki<5001> host=dynamic context=t***e nat=yes [5002] type=friend username=apu secret=password callerid=apu<5002> host=dynamic context=t***e nat=yes` I am writing my own sip client using jain-sip. The same configuration works well(bi-directional audio) when my sipclient and a 3rd party softphone are communication directly. It is only in the case of a conference call the audio from a 3rd party softphone to my sipclient isnt audible – Akshay Kasar Aug 09 '16 at 06:05
  • Sorry, nothing to add. Confbridge even not know that something like "nat" exist. Check sip.conf settings acordinly to nat guidlines. – arheops Aug 09 '16 at 13:11