[asterisk-bugs] [JIRA] Closed: (ASTERISK-20233) SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"

Matt Jordan (JIRA) noreply at issues.asterisk.org
Wed Aug 15 17:23:08 CDT 2012


     [ https://issues.asterisk.org/jira/browse/ASTERISK-20233?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan closed ASTERISK-20233.
----------------------------------

    Resolution: Not A Bug

> SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"
> -------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-20233
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20233
>             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
>
> Here is output
> v=0                                                                                                                                                                            
> o=<Private> 8001 8000 IN IP4 192.168.10.104                                                                                                                                    
> s=SIP Call                                                                                                                                                                     
> c=IN IP4 192.168.10.104                                                                                                                                                        
> t=0 0                                                                                                                                                                          
> m=audio 56008 RTP/SAVP 9 0 8 101                                                                                                                                               
> a=sendrecv                                                                                                                                                                     
> a=rtpmap:9 G722/8000                                                                                                                                                           
> a=ptime:20                                                                                                                                                                     
> a=rtpmap:0 PCMU/8000                                                                                                                                                           
> a=rtpmap:8 PCMA/8000                                                                                                                                                           
> a=rtpmap:101 telephone-event/8000                                                                                                                                              
> a=fmtp:101 0-15                                                                                                                                                                
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:txJVSUpEW30QbN7XrYuyOgNHVOHBX4dshzqYUwzg|2^31                                                                                        
> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:jcPz4ww9J2e7ONOZD4AkwronCQ8Jym6QNvXKz0jW|2^31                                                                                        
> <------------->                                                                                                                                                                
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: --- (17 headers 15 lines) ---                                                                                                  
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Sending to 109.237.252.179:51974 (NAT)                                                                                         
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Using INVITE request as basis request - 633650172-36867-4 at BJC.BGI.BA.BAE                                                       
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found peer '<private>' for '<private>' from xxx.xxx.xxx.xxx:51974                                                              
> [2012-08-15 17:33:27] VERBOSE[9911] netsock2.c:   == Using SIP RTP CoS mark 5                                                                                                  
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 9                                                                                                       
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 0                                                                                                       
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 8                                                                                                       
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found RTP audio format 101                                                                                                     
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format G722 for ID 9                                                                                   
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format PCMU for ID 0                                                                                   
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format PCMA for ID 8                                                                                   
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c: Found audio description format telephone-event for ID 101    
> [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:txJVSUpEW30QbN7XrYuyOgNHVOHBX4dshzqYUwzg|2^31       
> [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: SRTP crypto offer not acceptable                                                                                          
> [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: Crypto life time unsupported: crypto:2 AES_CM_128_HMAC_SHA1_32 inline:jcPz4ww9J2e7ONOZD4AkwronCQ8Jym6QNvXKz0jW|2^31       
> [2012-08-15 17:33:27] NOTICE[9911] sip/sdp_crypto.c: SRTP crypto offer not acceptable                                                                                          
> [2012-08-15 17:33:27] WARNING[9911] chan_sip.c: Can't provide secure audio requested in SDP offer                                                                              
> [2012-08-15 17:33:27] VERBOSE[9911] chan_sip.c:                                                                                                                                
> <--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:51974 --->                                                                                                                 
> SIP/2.0 488 Not acceptable here          
> [...]
> Call is ended with 488 error.     
> Same setup with blink softphone is OK. Difference is
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:j8ZJtC63YQGWyCMspHXEL6ca9VsuPcc2OBJk+Qav                                                                                             
> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:yxZP4ZTdJa8vcEV4FVQrkk/s7LLkjXlHeNzkCWWv
> So could the "|2^31" at the end of the crypto line be the cause ...
> -- 
> Daniel                                                             

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