[asterisk-bugs] [Asterisk 0018919]: Announced transfert with Aastra not works

Thu Mar 3 14:02:47 CST 2011

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18919 
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Reported By:                Bernard Merindol
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18919
Category:                   Channels/chan_sip/SRTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 309084 
Request Review:              
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Date Submitted:             2011-03-03 09:24 CST
Last Modified:              2011-03-03 14:02 CST
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Summary:                    Announced transfert with Aastra not works
Description: 
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

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---------------------------------------------------------------------- 
 (0132583) lmadsen (administrator) - 2011-03-03 14:02
 https://issues.asterisk.org/view.php?id=18919#c132583 
---------------------------------------------------------------------- 
<otherwiseguy> Could be a bug. They could post a pcap file as long as they
aren't also using TLS. In any case, probably something that we would just
try to reproduce and go from there. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-03-03 14:02 lmadsen        Note Added: 0132583                          
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