[asterisk-bugs] [JIRA] (ASTERISK-22748) SRTP Crypto Offer With Lifetime Not Accepted

[Michael L. Young (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22748?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Michael L. Young updated ASTERISK-22748:
----------------------------------------

    Summary: SRTP Crypto Offer With Lifetime Not Accepted  (was: SRTP Crypto Offer Not Acceptable)
    
> SRTP Crypto Offer With Lifetime Not Accepted
> --------------------------------------------
>
>                 Key: ASTERISK-22748
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22748
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP, Channels/chan_sip/TCP-TLS
>    Affects Versions: 11.5.1
>         Environment: FreePBX with Asterisk 11.5.1 recompiled
>            Reporter: Alejandro Mejia
>
> When a=crypto:1 and a=crypto:2 are not comming right after m=audio on SDP message from certain SIP clients (Grandstream phones for example), Asterisk ignores the crypto parameters and issues the following errors:
> NOTICE[20186][C-00000042]: sip/sdp_crypto.c:265 sdp_crypto_process: SRTP crypto offer not acceptable
> WARNING[20186][C-00000042]: chan_sip.c:10454 process_sdp: Rejecting secure audio stream without encryption details: audio 5004 RTP/SAVP 0 8 4 18 9 97 2 101
> This resulting on a "Not Acceptable Here" SIP error.
> The following SDP informations are from Yealink phone, and Grandstream phone.
> Yealink (call goes through without issues):
> v=0
> o=- 20013 20013 IN IP4 10.28.128.187
> s=SDP data
> c=IN IP4 10.28.128.187
> t=0 0
> m=audio 11792 RTP/SAVP 0 8 18 9 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NmU0NTlkM2QzNDkzNGFiNzVjYjE2MWI2ZDcyMWZk
> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:M2JhMmJmYmM4OGIxNDRlADY5NDQ5NjMANjljM2Qz
> a=crypto:3 F8_128_HMAC_SHA1_80 inline:Mzk2NDY1NWExYTdkYWI3YTdmOTc1MWZmNmRlYTkx
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:9 G722/8000
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=ptime:20
> a=sendrecv
> Grandstream phone (call won't go through):
> v=0
> o=898 8000 8000 IN IP4 10.28.128.97
> s=SIP Call
> c=IN IP4 10.28.128.97
> t=0 0
> m=audio 5004 RTP/SAVP 0 8 4 18 9 97 2 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:8 PCMA/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:9 G722/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:2 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Ar/jYxzGz1lLcROAnVi8IFGB2VJlynqKBhjaVvgb|2^32
> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CPvb7F73si5R/Z9kfT28OV0NujdfHwHaqQfyg13q|2^32

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