my paranoic side would recommend to hide/change private informations, specially any authentication line that might appear... this is certainly a sort of social engineering threat we should worry... better be safe than sorry....
On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <e...@uphreak.com> wrote: > I mean you can use a private gist, but you will be publishing the link in > a public email list. In general I personally dont believe revealing ip > addresses etc. is any problem - to put my money where my mouth is here is a > gist link to an unaltered SIP trace on my server :) > > https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 > > -Eric > > > On 06/23/2016 12:23 PM, John Nash wrote: > > Ok i am ready with logs. About gist may I use private option as traces > have our IPs, user > > On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <e...@uphreak.com> wrote: > >> Hey John, >> >> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >> from the proxy servers perspective and provide a link so that we can see >> what comes in, and what goes out from both sides. >> >> EG: ngrep -qtd any -W byline port 5060 >> >> This will show us the traffic that is leaving the proxy destined for the >> Freeswitch box, and what the freeswitch box sends back. >> >> Also - you can look in your browsers console log and provide the SIP >> trace from there in a seperate gist, so that we can see what opensips sends >> back up to your browser. >> >> -Eric >> >> >> Am I using correct sip.js example? I copied it to my server and accessing >> it using https: (used letsencrypt) >> >> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme < <e...@uphreak.com> >> e...@uphreak.com> wrote: >> >>> 1. I would suggest using SIP.js - <https://github.com/onsip/SIP.js> >>> https://github.com/onsip/SIP.js it is a much more active project that >>> sipml5. >>> >>> 2. Im guessing that you are not properly passing flags to RTPEngine. If >>> you want to have DTLS-SRTP between the browser, and plain RTP/AVP between >>> RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and >>> "answer" dtls-srtp back up to the browser. >>> >>> the offer to freeswitch would be: >>> >>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection >>> replace-origin ICE=remove"; >>> >>> >>> and the answer back up to the browswer would be: >>> >>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>> >>> >>> -Eric >>> >>> >>> >>> On 06/23/2016 08:20 AM, John Nash wrote: >>> >>> I am following >>> <http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2> >>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>> trying to test a call >>> >>> sipml5 ----------->Opensips + rtpengine --------> SIP end point >>> (Freeswitch) >>> >>> But I do not have any audio on both sides. I see this error at rtpengine >>> log "SRTP output wanted, but no crypto suite was negotiated" >>> >>> Anyone tested this scenario positive? >>> >>> >>> _______________________________________________ >>> Users mailing >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> _______________________________________________ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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