<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><br><div><div>On 26 Mar 2014, at 12:09, "Listas\" Nivaldo Montenegro Junior" <<a href="mailto:listasjr@digivox.com.br">listasjr@digivox.com.br</a>> <<a href="mailto:listasjr@digivox.com.br">listasjr@digivox.com.br</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><p dir="ltr">Hi, </p><p dir="ltr">So, It's not possible to implement using asterisk. The best solution is use a SIP Proxy? Using a SIP Proxy the internal calls (exten to exten) will not be processed by Asterisk. The SIP Proxy will handle that and Only calls that goes to PSTN will be handled by Asterisk, right? <br></p></blockquote>That depends on how you build the architecture. What you describe is one solution. </div><div><br></div><div>From a security standpoint, a server like Asterisk is a MITM-attack box. That's what a PBX is. If you want confidentiality in the calls between two phones, you do not involve a PBX unless you trust it a lot. </div><div><br></div><div>/O<br><blockquote type="cite"><p dir="ltr">
</p><p dir="ltr">Regards, <br></p><p dir="ltr">Nivaldo Montenegro Júnior </p>
<div class="gmail_quote">Em 26/03/2014 03:48, "Olle E. Johansson" <<a href="mailto:oej@edvina.net">oej@edvina.net</a>> escreveu:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word"><br><div><div>On 26 Mar 2014, at 02:41, "Listas\" Nivaldo Montenegro Junior" <<a href="mailto:listasjr@digivox.com.br" target="_blank">listasjr@digivox.com.br</a>> <<a href="mailto:listasjr@digivox.com.br" target="_blank">listasjr@digivox.com.br</a>> wrote:</div>
<br><blockquote type="cite"><div dir="ltr"><div style="font-family:Tahoma,Verdana;font-size:13px;text-align:-webkit-auto"><font face="Tahoma">Hi,</font></div><div style="font-family:Tahoma,Verdana;font-size:13px;text-align:-webkit-auto">
<font face="Tahoma"><br></font></div><div style="font-family:Tahoma,Verdana;font-size:13px;text-align:-webkit-auto"><font face="Tahoma">We are looking for a developer or asterisk consultant to implement the direct media setup on Asterisk using TLS and SRTP.</font></div>
<div style="font-family:Tahoma,Verdana;font-size:13px;text-align:-webkit-auto"><font face="Tahoma">We tested using SIP + RTP and it works fine with the parameter directmedia=yes. But when we enable the SRTP, this stops to work.</font></div>
</div></blockquote>For SRTP to work peer to peer you need to have the key exchange either going through Asterisk or in the media stream, like DTLS. It may be easier to use a SIP proxy to get the key exchange to work properly. </div>
<div><br></div><div>To facilitate SDES key exchange through Asterisk involves a lot of issues, which from a security standpoint are rather scary. Asterisk will offer someone else's keys and need to switch back and forth during the call.</div>
<div><br></div><div>Anything can be done though, it's just source code. :-)</div><div><br></div><div>/O<br><blockquote type="cite"><div dir="ltr">
<div style="font-family:Tahoma,Verdana;font-size:13px;text-align:-webkit-auto"><div><br></div><div>We are very interested in this project and we will pay for it.</div><div><br></div>
<div>If any one has interest on this project, please send me an e-mail to <a href="mailto:junior@digivox.com.br" target="_blank">junior@digivox.com.br</a>.</div><div><br></div><div><br></div><div>Thanks,</div>
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<strong><span lang="EN-US" style="font-family:verdana">Nivaldo Montenegro Júnior</span></strong><span lang="EN-US" style="font-family:verdana"> </span><b><i><span lang="EN-US" style="font-family:verdana;font-size:7.5pt"><br>
</span></i></b><em><b><span lang="EN-US" style="font-family:verdana;font-size:7.5pt">Diretor de TI&C</span></b></em><b><i><span lang="EN-US" style="font-family:verdana;font-size:7.5pt"><br></span></i></b><b><span lang="EN-US" style="font-family:verdana;color:blue;font-size:7.5pt">Digivox Soluções de Comunicação Ltda<br>
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