Displaying 4 results from an estimated 4 matches for "local_key64".
2015 Apr 17
1
Asterisk 11 SRTP: unsupported crypto parameters: UNENCRYPTED_SRTCP
...50096 RTP/SAVP 0 18 120
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:120 telephone-event/8000
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4BkOwcCkD/AjM|2^20 UNENCRYPTED_SRTCP
And on CLI I see,
DEBUG[1568][C-00000000] sip/sdp_crypto.c: local_key64
7vXot5kn/sl/GYv5ENN6yW0PZZapQ00c++biLgoX len 40
WARNING[1568][C-00000000] sip/sdp_crypto.c: Unsupported crypto parameters:
UNENCRYPTED_SRTCP
DEBUG[1568][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4BkOwcCkD/AjM|2^20...
2010 Dec 24
5
SRTP unprotect: authentication failure
...hen the same warning again.
Asterisk 1.8.1.1, RealTime engine, sip peer has encrytion->yes
The client program is CSipSimple on Android
Here are some log file traces:
Peer 0010101 is calling some number that is routed to context a2billing
[2010-12-23 11:06:22] DEBUG[5941] sip/sdp_crypto.c: local_key64 3gWGFJAffj4Pn393BUPwe3/wOMx5/ndZyPtfno7L len 40
[2010-12-23 11:06:22] DEBUG[5941] sip/sdp_crypto.c: SRTP policy activated
[2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0VyG/fnup0U9qDoTGlWvVuE5yAef5MfYU6F67oI+... OK.
[2010-...
2020 Jun 08
0
pjsip extensions rings but call drop on answer
...9] DEBUG[4180] netsock2.c: Splitting 'asterisk2' into...
[Jun 8 12:28:09] DEBUG[4180] netsock2.c: ...host 'asterisk2' and port ''.
[Jun 8 12:28:09] DEBUG[4180] res_rtp_asterisk.c: Setup RTCP on RTP
instance '0x7f05780618e0'
[Jun 8 12:28:09] DEBUG[4180] res_srtp.c: local_key64
2Rbo7TRiuRAnS0IYJeSn0ELEYAVnkOVCUwou7pxO len 40
[Jun 8 12:28:09] DEBUG[4180] res_pjsip_sdp_rtp.c: Stream msid:
0x7f0578077610 audio 23eb03ca-f0ee-406a-b7cd-5fb19fc33fa2
ddca7927-ff8d-45ab-a61f-9474f8b7a9df
[Jun 8 12:28:09] DEBUG[4180] res_pjsip_session.c: Method is INVITE
[Jun 8 12:28:09] DEBUG[...
2012 Aug 17
2
How to test Websocket support in SIP in Asterisk trunk?
I see no indication of how to do this in sip.conf, and when I start
Asterisk, it doesn't wait on port 80.
Greetings,
--
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regi?es: (11)4063-6100