SIP NAT

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In order to solve the common problems with lossing voice when passing through NAT, YATE uses the RTP stream's IP+Port instead of the RTP IP+Port declared in SDP in the ''200 OK'' response. This small hack allows most of the SIP CLIENTS when Yate is the SERVER to pass the voice over NAT.  
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In order to solve the common problems with lossing voice when passing through NAT, YATE uses the RTP stream's IP+Port instead of the RTP IP+Port declared in SDP in the ''200 OK'' response.  
  
[[File:SIP_NAT.png]]
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This small hack allows most of the SIP CLIENTS when Yate is the SERVER to pass the voice over NAT.  
  
 
A better explanation can be found at: http://freshmeat.net/articles/view/2079/
 
A better explanation can be found at: http://freshmeat.net/articles/view/2079/
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[[File:SIP_NAT.png]]
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'''See also'''
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* [http://freshmeat.net/articles/view/2079/ NAT traversal for the SIP protocol]
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* [[SIP_in_Yate#SIP_with_NAT_in_Yate|SIP with NAT in Yate]]
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* [[Telephony]]
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[[Category:SIP]] [[Category:NAT]] [[Category:Server]]

Latest revision as of 11:04, 4 November 2013

In order to solve the common problems with lossing voice when passing through NAT, YATE uses the RTP stream's IP+Port instead of the RTP IP+Port declared in SDP in the 200 OK response.

This small hack allows most of the SIP CLIENTS when Yate is the SERVER to pass the voice over NAT.

A better explanation can be found at: http://freshmeat.net/articles/view/2079/

SIP NAT.png


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