Dmytro - Thank you for your reply. Adding additional relays or proxies will add two big times to latency. First there is the added latency of the relay or proxy connection. Then there is the added latency of the sip client adding redundant layers of media stream encryption to prevent MitM at the relay/proxy. If the connection is direct p2p between the two hidden service sip clients then you can use tor for the media stream encryption while maintaining the same level of media stream security as say a direct ZRTP connection. Therefore non-tor relays (like TURN) or proxies can be used and would also not be required as long as pjsip speaks TCP for signalization and media stream. Thanks! Aaron On Sat, Apr 12, 2014 at 12:50 AM, Dmytro Bogovych <dmytro.bogovych at gmail.com > wrote: > AFAIK it can do TURN relaying via TLS connections. > > > On Fri, Apr 11, 2014 at 10:51 PM, Aaron Lux <a at aaronlux.com> wrote: > >> Has a tcp transport adapter has been developed for pjsip? >> >> I'm interested in bringing features of speakfreely[1] to android users >> using an open protocol instead of the proprietary speakfreely protocol. >> R?gis from CSipSimple was kind enough to locate this[3] ~6 year old post >> from Benny making it seem very likely work has been done to bring >> rfc4571[2] to pjsip. >> >> I am working on a proof of concept showing audio can be passed over TCP >> between two Android Orbot clients running CSipSimple on a hidden service >> without a lot of latency. >> >> This is important for user privacy since it reduces or totally eliminates >> trap & trace[4]. In the real world "trap and trace" is the biggest >> privacy threat to almost all phone users because it is easiest to analyze. >> In addition, trap & trace may cause problems for purveyors of private >> communications since they are just one subpoena away from losing their >> credibility or even their business/freedom. >> >> Thanks! >> Aaron >> >> [1] http://torfone.org/spfr.html >> [2] https://tools.ietf.org/html/rfc4571 >> [3] >> http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003204.html >> [4] When (time of call, length of call), Where (location of IP), and Who >> (Identity of caller and callee). With this information call content can be >> derived using basic investigation and interviewing techniques even when >> content was encrypted. >> >> instead of udp because >> >> >> and wondering if anyone has already developed >> >> so I can make voice calls over Orbot on Android >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140412/a93f5731/attachment.html>
