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

Mon Mar 7 07:19:06 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:                       SWP-3205 
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:              2011-03-07 07:19 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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---------------------------------------------------------------------- 
 (0132686) Bernard Merindol (reporter) - 2011-03-07 07:19
 https://issues.asterisk.org/view.php?id=18919#c132686 
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I have added files.
asterisk IP: 192.168.169.60
TelA P1000 IP 192.168.169.102
TelB P1001 IP 192.168.169.100
TelC P1002 IP 192.168.169.101

Sory for this IP addresses  is my DHCP. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-03-07 07:19 Bernard MerindolNote Added: 0132686                          
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