[asterisk-dev] Asterisk 11.3.0-rc1 and srtp - white noise only

Wed Mar 20 10:41:04 CDT 2013

Quick follow-up, I believe that recent changes related to sdp_crypto are causing the issue.

Here is another log, Call Flow 

Gigaset w/o srtp > Asterisk > snom.

Look at the crypto logging. When Asterisk is processing the remote SDP answer, it is logging his own key and not the one from SDP - I assume that it is then trying to decode the remote srtp stream with the wrong key, and not with the proper remote from the SDP. This would explain the white noise.

[Mar 20 16:31:24] VERBOSE[13764][C-00000002] netsock2.c:   == Using SIP RTP CoS mark 5
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] pbx.c:     -- Executing [1 at local:1] Dial("SIP/1939.N720-00000004", "sip/snom360.2") in new stack
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] netsock2.c:   == Using SIP RTP CoS mark 5
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] chan_sip.c: Audio is at 14370
[Mar 20 16:31:24] DEBUG[13805][C-00000002] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:cEglQBq1wgUwFUV6Wg++6QzqZ0tUlSmA1hZSkmhE
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] chan_sip.c: Adding codec 100012 (g722) to SDP
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.36.107:2168:
INVITE sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o SIP/2.0
Via: SIP/2.0/TLS 192.168.32.71:5061;branch=z9hG4bK07093a26
Max-Forwards: 70
From: "Crazy Chrissy" <sip:4 at 192.168.32.71>;tag=as4e664e1f
To: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>
Contact: <sip:4 at 192.168.32.71:5061;transport=TLS>
Call-ID: 6abf09b90dc59e9c056447de3e46c608 at 192.168.32.71:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.3.0-rc1
Date: Wed, 20 Mar 2013 15:31:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: "Crazy Chrissy" <sip:4 at 192.168.32.71>
Content-Type: application/sdp
Content-Length: 350

v=0
o=root 1039580908 1039580908 IN IP4 192.168.32.71
s=Asterisk PBX 11.3.0-rc1
c=IN IP4 192.168.32.71
t=0 0
m=audio 14370 RTP/SAVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:cEglQBq1wgUwFUV6Wg++6QzqZ0tUlSmA1hZSkmhE

---
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] app_dial.c:     -- Called sip/snom360.2
[Mar 20 16:31:24] VERBOSE[13795] chan_sip.c:
<--- SIP read from TLS:192.168.36.107:2168 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.32.71:5061;branch=z9hG4bK07093a26
From: "Crazy Chrissy" <sip:4 at 192.168.32.71>;tag=as4e664e1f
To: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>;tag=iwo9v6z5oo
Call-ID: 6abf09b90dc59e9c056447de3e46c608 at 192.168.32.71:5061
CSeq: 102 INVITE
Contact: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>;reg-id=1
Content-Length: 0

<------------->
[Mar 20 16:31:24] VERBOSE[13795] chan_sip.c: --- (8 headers 0 lines) ---
[Mar 20 16:31:24] VERBOSE[13795] chan_sip.c:
<--- SIP read from TLS:192.168.36.107:2168 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.32.71:5061;branch=z9hG4bK07093a26
From: "Crazy Chrissy" <sip:4 at 192.168.32.71>;tag=as4e664e1f
To: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>;tag=iwo9v6z5oo
Call-ID: 6abf09b90dc59e9c056447de3e46c608 at 192.168.32.71:5061
CSeq: 102 INVITE
Contact: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

<------------->
[Mar 20 16:31:24] VERBOSE[13795] chan_sip.c: --- (10 headers 0 lines) ---
[Mar 20 16:31:24] VERBOSE[13795][C-00000002] chan_sip.c: list_route: hop: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>
[Mar 20 16:31:24] VERBOSE[13805][C-00000002] app_dial.c:     -- SIP/snom360.2-00000005 is ringing
[Mar 20 16:31:27] VERBOSE[13795] chan_sip.c:
<--- SIP read from TLS:192.168.36.107:2168 --->
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.32.71:5061;branch=z9hG4bK07093a26
From: "Crazy Chrissy" <sip:4 at 192.168.32.71>;tag=as4e664e1f
To: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>;tag=iwo9v6z5oo
Call-ID: 6abf09b90dc59e9c056447de3e46c608 at 192.168.32.71:5061
CSeq: 102 INVITE
Contact: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>;reg-id=1
User-Agent: snom360/8.4.35
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 351

v=0
o=root 1601234252 1601234253 IN IP4 192.168.36.107
s=call
c=IN IP4 192.168.36.107
t=0 0
m=audio 12780 RTP/SAVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fMswsjlBRMou3kub16WuCuVQTcCcM9E4pXxu+JqW
a=direction:both
a=sendrecv
<------------->
[Mar 20 16:31:27] VERBOSE[13795] chan_sip.c: --- (13 headers 14 lines) ---
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Found RTP audio format 9
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Found RTP audio format 8
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Found RTP audio format 101
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Found audio description format G722 for ID 9
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 101
[Mar 20 16:31:27] DEBUG[13795][C-00000002] sip/sdp_crypto.c: Accepting crypto tag 1
[Mar 20 16:31:27] DEBUG[13795][C-00000002] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:cEglQBq1wgUwFUV6Wg++6QzqZ0tUlSmA1hZSkmhE
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Capabilities: us - (alaw|g722), peer - audio=(alaw|g722)/video=(nothing)/text=(nothing), combined - (alaw|g722)
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Peer audio RTP is at port 192.168.36.107:12780
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: list_route: hop: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: set_destination: Parsing <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o> for address/port to send to
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: set_destination: set destination to 192.168.36.107:2168
[Mar 20 16:31:27] VERBOSE[13795][C-00000002] chan_sip.c: Transmitting (no NAT) to 192.168.36.107:2168:
ACK sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o SIP/2.0
Via: SIP/2.0/TLS 192.168.32.71:5061;branch=z9hG4bK657c1ced
Max-Forwards: 70
From: "Crazy Chrissy" <sip:4 at 192.168.32.71>;tag=as4e664e1f
To: <sip:snom360.2 at 192.168.36.107:2168;transport=tls;line=t3uroc3o>;tag=iwo9v6z5oo
Contact: <sip:4 at 192.168.32.71:5061;transport=TLS>
Call-ID: 6abf09b90dc59e9c056447de3e46c608 at 192.168.32.71:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.3.0-rc1
Content-Length: 0

---
[Mar 20 16:31:27] VERBOSE[13805][C-00000002] app_dial.c:     -- SIP/snom360.2-00000005 answered SIP/1939.N720-00000004
[Mar 20 16:31:29] VERBOSE[13764] chan_sip.c: Really destroying SIP dialog '27bb4e8849db8ba764e42ae45b433b23 at 192.168.32.71:5061' Method: BYE

Any ideas on this?

Best regards,
Martin
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