[asterisk-bugs] [JIRA] Updated: (ASTERISK-20234) SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"

Rusty Newton (JIRA) noreply at issues.asterisk.org
Tue Aug 21 09:41:07 CDT 2012


     [ https://issues.asterisk.org/jira/browse/ASTERISK-20234?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-20234:
------------------------------------

    Description: 
As you can see, snom 320 (latest stable firmware snom320-SIP 8.7.3.10) is annoncing crypto but asterisk doesn't recognize it.

[Edit by Rusty Newton - removed debug excerpt since full log is now attached]

And call is not accepted

-- 
Daniel

  was:
As you can see, snom 320 (latest stable firmware snom320-SIP 8.7.3.10) is annoncing crypto but asterisk doesn't recognize it.

v=0                                                                                                                                                                            
o=root 80443371 80443371 IN IP4 192.168.10.105                                                                                                                                 
s=call                                                                                                                                                                         
c=IN IP4 192.168.10.105                                                                                                                                                        
t=0 0                                                                                                                                                                          
m=audio 49154 RTP/AVP 9 0 8 3 99 108 18 101                                                                                                                                    
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:nCj5W+omFyJ2qyncFHRWrUBcmSSOVXAs7E9xQy7x  ;<- here
a=rtpmap:9 G722/8000                                                                                                                                                           
a=rtpmap:0 PCMU/8000                                                                                                                                                           
a=rtpmap:8 PCMA/8000                                                                                                                                                           
a=rtpmap:3 GSM/8000                                                                                                                                                            
a=rtpmap:99 G726-32/8000                                                                                                                                                       
a=rtpmap:108 AAL2-G726-32/8000                                                                                                                                                 
a=rtpmap:18 G729/8000                                                                                                                                                          
a=fmtp:18 annexb=no                                                                                                                                                            
a=rtpmap:101 telephone-event/8000                                                                                                                                              
a=fmtp:101 0-15                                                                                                                                                                
a=ptime:20                                                                                                                                                                     
a=sendrecv                                                                                                                                                                     
<------------->                                                                                                                                                                
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: --- (20 headers 19 lines) ---                                                                                                  
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Sending to xxx.xxx.xxx.xxx:2048 (NAT)                                                                                          
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Using INVITE request as basis request - 502b8cb5f0cb-gq8fmvfprog9                                                              
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found peer 'private' for 'private' from xxx.xxx.xxx.xxx:2048                                                               
[2012-08-15 15:50:20] VERBOSE[9001] netsock2.c:   == Using SIP RTP CoS mark 5                                                                                                  
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 9                                                                                                       
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 0                                                                                                       
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 8                                                                                                       
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 3                                                                                                       
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 99                                                                                                      
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 108                                                                                                     
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 18                                                                                                      
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 101                                                                                                     
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format G722 for ID 9                                                                                   
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format PCMU for ID 0                                                                                   
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format PCMA for ID 8                                                                                   
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format GSM for ID 3                                                                                    
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format G726-32 for ID 99                                                                               
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format AAL2-G726-32 for ID 108                                                                         
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format G729 for ID 18                                                                                  
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format telephone-event for ID 101                                                                      
[2012-08-15 15:50:20] WARNING[9001] chan_sip.c: We are requesting SRTP for audio, but they responded without it! ;???
[2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c:                                                                                                                                
<--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:2048 --->          
SIP/2.0 488 Not acceptable here

And call is not accepted

-- 
Daniel


> SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
> ----------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-20234
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20234
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 10.7.0
>         Environment: RHEL5 linux 2.6.18-308.11.1.el5
>            Reporter: tootai
>            Assignee: tootai
>         Attachments: full
>
>
> As you can see, snom 320 (latest stable firmware snom320-SIP 8.7.3.10) is annoncing crypto but asterisk doesn't recognize it.
> [Edit by Rusty Newton - removed debug excerpt since full log is now attached]
> And call is not accepted
> -- 
> Daniel

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