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</o:shapelayout></xml><![endif]--></head><body lang=SV link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-US>Hello,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I’m trying to setup an Asterisk (version 1.8.8) to do SRTP termination and then send the call on to other servers, unencrypted. All the basics work fine.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I want the Asterisk to do as little as possible with the RTP packets and no transcoding. We always make sure to force same codec on incoming and outgoing call leg.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>When not using SRTP, Asterisk does P2P bridging of the RTP packets. That is, simply copying the packets, which is the expected result. But when we send in SRTP media, Asterisk starts decode/encode voice data instead of just do P2P bridging. <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I also notice Asterisk doesn’t say “Locally bridging channels” in the latter case, which might be the clue that we’re not doing P2P bridging.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Why can we not use P2P bridging when doing SRTP->RTP media conversion? Is there anything we can change in the source code to force packet bridging in this case?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Best regards,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Jan Blom<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p></div></body></html>