mancyborg at gmail.com
2010-Feb-24 21:05 UTC
[asterisk-users] Encrypted calls between mobile gsm and isdn (asterisk)
Hi All, are you aware of any solution which can encrypt calls between a mobile gsm and isdn (asterisk) ? Thanks for your attention, have a nice day. Mike
Hello, I have a setup that includes a cellphone a proxy running Kamailio and rtpproxy and a SIP server (VoipSwitch/Asterisk). Call flow works well while using Asterisk, however when VoipSwitch is used i find that the BYE message from VoipSwitch has an RURI = account at VoipSwitch, so the proxy ends up repeatedly sending BYE messages to VoipSwitch instead of sending them to the Cellphone, causing the Cellphone to never hangup. However when using Asterisk the BYE message is forwarded to the cellphone and both endpoints of the call hangup. I show below the SIP message flow while using VoipSwitch. Cell Phone Kamailio VoipSwitch | | | |INVITE | | |------------->| | |100 Trying | | |<-------------| | | |INVITE | | |------------->| | |100 trying | | |<-------------| | |183SessionProg| | |<-------------| |183SessionProg| | |<-------------| | | | 200 OK | | 200 OK |<-------------| |<-------------| | | ACK | | |------------->| | | | ACK | | |------------->| | | BYE | | |<-------------|<- BYE,RURI=account at VoipSwitch | | BYE | | |------------->| | | BYE | | |------------->| Is this issue caused by the SIP server or some other element along the SIP message flow ? Does anybody know the difference in SIP message handling between VoipSwitch and Asterisk or can anybody point me to an online resource ? -- Thanks and Regards, Vikram Ragukumar.