Re: [Linphone-developers] SRTP and updateCall

[Michael S]

Ok, I found solution:

I have to set 
params.setMediaEnctyption(LinphoneCore.MediaEncryption.SRTP)
and pass params into updateCall(call, params) 

14.02.2017, 15:22, "Michael S" <address@hidden>:
> Hi,
>
> We are using SRTP and we were able to get this working with audio, but failed 
> with video.
>
> For video calls, we are calling lc.updateCall() when camera turned on or 
> switched front/back.
>
> updateCall() leads to INVITE with "Subject: Media changed" but liblinphone 
> forgot about a=crypto attributes.
>
> As result, since SIP proxy server configured with mandatory SRTP, server 
> responds with 488.
>
> For example, initial INVITE
>
>    INVITE sip:address@hidden SIP/2.0
>    Via: SIP/2.0/TLS 172.17.254.7:46160;branch=z9hG4bK.IKYuBZnFr;rport
>    From: <sip:address@hidden>;tag=e~bm1pyC2
>    To: sip:address@hidden
>    CSeq: 20 INVITE
>    Call-ID: RC1cHuZLOb
>    Max-Forwards: 70
>    Supported: replaces, outbound
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO, UPDATE
>    Content-Type: application/sdp
>    Content-Length: 1210
>    Contact: 
> <sip:address@hidden:46160;transport=tls>;+sip.instance="<urn:uuid:bf87b5fd-20d9-4613-87bd-d5ef4a5ab559>"
>    User-Agent: Unknown (belle-sip/1.5.0)
>
>    v=0
>    o=111111 568 153 IN IP4 172.17.254.7
>    s=Talk
>    c=IN IP4 172.17.254.7
>    t=0 0
>    a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
>    m=audio 7078 RTP/SAVPF 96 101
>    a=rtpmap:96 SILK/8000
>    a=rtpmap:101 telephone-event/8000
>    a=crypto:1 AES_CM_128_HMAC_SHA1_80 
> inline:XiIh1YhpXxRYTdanGEKZRqfbo/+i9ctxVL8I8hnN
>    a=crypto:2 AES_CM_128_HMAC_SHA1_32 
> inline:z+8cgOd6LbhI2v1HF2kXxceg/3PELL5ea+N6Phyl
>    a=crypto:3 AES_256_CM_HMAC_SHA1_80 
> inline:OAKH+Y7jiFJ3OSI0zRcqjpWEX0ZWk6RzdNPUbigNCUBPKuHeIMcdBpfOo1s76g==
>    a=crypto:4 AES_256_CM_HMAC_SHA1_32 
> inline:l26+a3G8HAh2D2re3cQ02O3d4O3D2JeyQQvVWAXcN+NXsQa2/1OzEtM75JU31Q==
>    a=rtcp-fb:* trr-int 5000
>    m=video 9078 RTP/SAVPF 96
>    a=rtpmap:96 VP8/90000
>    a=crypto:1 AES_CM_128_HMAC_SHA1_80 
> inline:3SNxSjXVQsLqUPFCBDrAtyFcrUYCmP0ANL8aWUcX
>    a=crypto:2 AES_CM_128_HMAC_SHA1_32 
> inline:s+8vqzhpK6smHtEYLtkiCIWjPXoj0buWrBzEtmUl
>    a=crypto:3 AES_256_CM_HMAC_SHA1_80 
> inline:XvtzoDTyUpmiqKYHummTsH54JWszOSrzAADPHXXma0Wdy1MkPF9CDaIYNXpd5Q==
>    a=crypto:4 AES_256_CM_HMAC_SHA1_32 
> inline:3goEB6m0TOgcmGNs6n3bgHd9Gh5J1JfIPFStjiCMLRTrNm31IHdJj8lP6WYlsQ==
>    a=rtcp-fb:* trr-int 5000
>    a=rtcp-fb:96 nack pli
>    a=rtcp-fb:96 nack sli
>    a=rtcp-fb:96 ack rpsi
>    a=rtcp-fb:96 ccm fir
>
> And INVITE, caused by updateCall()
>
>    INVITE sip:address@hidden:55061;transport=tls SIP/2.0
>    Via: SIP/2.0/TLS 172.17.254.5:46595;branch=z9hG4bK.~uiob3qVr;rport
>    From: <sip:address@hidden>;tag=xf4Iqg~
>    To: "+111111" <sip:address@hidden>;tag=9FF3pyeZQFrBg
>    CSeq: 111 INVITE
>    Call-ID: 0de0e1db-6cb9-1235-48b9-6cae8b3b6e92
>    Max-Forwards: 70
>    Subject: Media change
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO, UPDATE
>    Content-Type: application/sdp
>    Content-Length: 295
>    Contact: 
> <sip:address@hidden:46595;transport=tls>;expires=604800;+sip.instance="<urn:uuid:89df6d19-8b7e-4b78-abe7-5e0793bd8865>"
>    User-Agent: Unknown (belle-sip/1.5.0)
>
>    v=0
>    o=222222 570 103 IN IP4 172.17.254.5
>    s=Talk
>    c=IN IP4 172.17.254.5
>    t=0 0
>    a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
>    m=audio 7078 RTP/AVP 96 101
>    a=rtpmap:96 SILK/8000
>    a=rtpmap:101 telephone-event/8000
>    m=video 9078 RTP/AVP 96
>    a=rtpmap:96 VP8/90000
>
> And server response:
>
> 389347c6-e4b2-4436-bd2a-d7898e04c1f3 2017-02-13 23:59:56.765808 [DEBUG] 
> switch_core_media.c:4843 Video Codec Compare [VP8:96]/[VP8:99]
> 389347c6-e4b2-4436-bd2a-d7898e04c1f3 2017-02-13 23:59:56.765808 [DEBUG] 
> switch_core_media.c:4885 Video Codec Compare [VP8:96] +++ is saved as a match
> 389347c6-e4b2-4436-bd2a-d7898e04c1f3 2017-02-13 23:59:56.765808 [WARNING] 
> switch_core_media.c:4901 Crypto not negotiated but required.
> 389347c6-e4b2-4436-bd2a-d7898e04c1f3 2017-02-13 23:59:56.765808 [ERR] 
> sofia.c:7884 Reinvite Codec Error!
> send 563 bytes to tls/[202.202.202.202]:46595 at 23:59:56.773338:
>    ------------------------------------------------------------------------
>    SIP/2.0 488 Not Acceptable Here
>    Via: SIP/2.0/TLS 
> 172.17.254.5:46595;branch=z9hG4bK.~uiob3qVr;rport=46595;received=202.202.202.202
>    From: <sip:address@hidden>;tag=xf4Iqg~
>    To: "+111111" <sip:address@hidden>;tag=9FF3pyeZQFrBg
>    Call-ID: 0de0e1db-6cb9-1235-48b9-6cae8b3b6e92
>    CSeq: 111 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.6.13-21-e755b43~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, 
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
> Please help.
>
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