Summary: | ASTERISK-11016: About SRTP connection error | ||
Reporter: | Zuo-Ren, Liou (oarpvfpre) | Labels: | |
Date Opened: | 2007-12-12 04:15:05.000-0600 | Date Closed: | 2007-12-12 09:12:23.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Core/RTP |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Hi there, I have successfully compiled SVN-trunk-r81432M with ast_srtp_r81432_mikey_r3412.patch and connected two minisip. A secure call to the number 01 or the echo test works nice. When I try to make a secure call between the two phones, Asterisk show a message "Cannot native bridge in SRTP". I have try to set "canreinvite=no" in sip.conf, but it still doesn't work, what is the next step to enable secure calls between two phones? ****** ADDITIONAL INFORMATION ****** -------------------------------------- configuration in sip.conf -------------------------------------- [isec_01] type=friend username=isec_01 secret=isec host=dynamic context=tutorial canreinvite=no nat=no -------------------------------------- configuration in extensions.conf -------------------------------------- [tutorial] exten => 01, 1, Set(_SIP_SRTP_SDES=1) exten => 01, n, Set(_SIPSRTP=optional) exten => 01, n, Set(_SIPSRTP_CRYPTO=enable) exten => 01, n, Dial(SIP/isec_01) -------------------------------------- message in Asterisk CLI -------------------------------------- [Dec 12 17:50:04] NOTICE[2562]: sdp_mikey.c:112 sdp_mikey_setup: Using MIKEY PSK isec -- Executing [01@tutorial:1] Set("SIP/isec_02-08374fc0", "_SIP_SRTP_SDES=1") in new stack -- Executing [01@tutorial:2] Set("SIP/isec_02-08374fc0", "_SIPSRTP=optional") in new stack -- Executing [01@tutorial:3] Set("SIP/isec_02-08374fc0", "_SIPSRTP_CRYPTO=enable") in new stack -- Executing [01@tutorial:4] Dial("SIP/isec_02-08374fc0", "SIP/isec_01") in new stack == Using TOS bits 0 == Using CoS mark 5 [Dec 12 17:50:04] NOTICE[2562]: chan_sip.c:3554 sip_call: SIPSRTP_CRYPTO [Dec 12 17:50:04] NOTICE[2562]: chan_sip.c:3541 sip_call: SIPSRTP -- Called isec_01 -- SIP/isec_01-0837e730 is ringing --- set_address_from_contact host '140.125.84.30' -- SIP/isec_01-0837e730 answered SIP/isec_02-08374fc0 [Dec 12 17:50:08] NOTICE[2562]: rtp.c:3924 ast_rtp_bridge: Cannot native bridge in SRTP. | ||
Comments: | By: Joshua C. Colp (jcolp) 2007-12-12 09:12:23.000-0600 Opening a new bug for an outside patch is not what the bug tracker is for, if a bug already exists please add your comments there. |