[asterisk-bugs] [JIRA] Commented: (ASTERISK-17505) Announced transfert with Aastra not works
Bernard Merindol (JIRA)
noreply at issues.asterisk.org
Thu Jul 26 15:17:21 CDT 2012
[ https://issues.asterisk.org/jira/browse/ASTERISK-17505?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=195256#comment-195256 ]
Bernard Merindol commented on ASTERISK-17505:
---------------------------------------------
Hi Matt,
Thank for this comment.
This issue is close for me. From version 1.8.11 the re-invite works fine in SRTP. I have removed my patch and use the normal Asterisk for my customers.
Thank and best regards
Bernard
> Announced transfert with Aastra not works
> -----------------------------------------
>
> Key: ASTERISK-17505
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-17505
> Project: Asterisk
> Issue Type: Bug
> Components: Channels/chan_sip/SRTP
> Affects Versions: 1.8.3
> Reporter: Bernard Merindol
> Severity: Minor
> Attachments: full, res_srtp.c.patch, srtppb.pcap
>
>
> Hi,
> When use SRTP with aastra phone the announced transfert not works.
> A call B, B call C for prepare transfert, C accept transfert,B finish transfer. C ear A, but A not ear C.
> For me this problem is due at C send new crypto key in OK of (re)-Invite. In this cas asterisk not change the key and the RTC traffic from C to a is not uncrypted by Asterisk.
> In full I see:
> [Mar 3 16:17:54] WARNING[10455] res_srtp.c: SRTP unprotect: authentication failure
> [Mar 3 16:17:54] WARNING[10455] res_srtp.c: SRTP unprotect: authentication failure
> [Mar 3 16:17:54] WARNING[10455] res_srtp.c: SRTP unprotect: authentication failure
> Before I see:
> IN OK of aastra (tne C phone)
> --- SIP read from TLS:192.168.169.214:5061 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/TLS 192.168.169.60:5061;branch=z9hG4bK650a3a5b;rport=5061;received=192.168.169.60
> From: "P1001" <sip:1001 at 192.168.169.60>;tag=as18df4bfc
> To: <sips:1002 at 192.168.169.214:5061>;tag=1795192984
> Call-ID: 402518d63295ba81158bf5584f87abc3 at 192.168.169.60:5061
> CSeq: 103 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
> Allow-Events: talk, hold, conference, LocalModeStatus
> Contact: "TCE" <sips:1002 at 192.168.169.214:5061>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D28CD53>"
> Require: timer
> Server: Aastra 6731i/2.6.0.2010
> Session-Expires: 900;refresher=uas
> Supported: gruu, path, timer, replaces
> Content-Type: application/sdp
> Content-Length: 297
>
> v=0
> o=MxSIP 0 1 IN IP4 192.168.169.214
> s=SIP Call
> c=IN IP4 192.168.169.214
> t=0 0
> m=audio 8000 RTP/SAVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjd2dixXKzBlQ3okdUpGK0IwZHB3Y1lGIXtKJT45
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> The asterisk process:
> [Mar 3 16:16:11] DEBUG[10255] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
> [Mar 3 16:16:11] DEBUG[10255] res_srtp.c: Policy already exists, not re-adding
> [Mar 3 16:16:11] WARNING[10255] sip/sdp_crypto.c: Could not set local SRTP policy
> [Mar 3 16:16:11] DEBUG[10255] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjd2dixXKzBlQ3okdUpGK0IwZHB3Y1lGIXtKJT45... UNSUPPORTE\
> D.
> [Mar 3 16:16:11] DEBUG[10255] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
> [Mar 3 16:16:11] DEBUG[10255] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
> [Mar 3 16:16:11] DEBUG[10255] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
> In this case Asterisk not change the policy.
> I have tested with iPHONE with BRIA sip phone in C phone. The transfert works fine. But Bria not change the crypto in OK.
> Thank for your help.
> Best regards
--
This message is automatically generated by JIRA.
For more information on JIRA, see: http://www.atlassian.com/software/jira
More information about the asterisk-bugs
mailing list