[asterisk-bugs] [JIRA] (ASTERISK-20905) Video returned in SIP SDP RTP/AVP when it should be RTP/SAVP when SRTP (encryption=yes) is enabled.
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Fri Jan 11 15:29:45 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-20905?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-20905:
------------------------------------
Reference Notes: [Edit by Rusty Newton - removed debug and attached as log.txt as per the guidelines] (was: Here's the full log with verbose+debug set to max;
The SIP call started with only Audio, then transitioned to Video. Extension 777 was running the Echo application to provide video loopback. When encryption=no, this exact process functions properly with videoi.
<--- SIP read from TLS:xxx.xxx.xxx.52:60099 --->
INVITE sip:777 at xxx.xxx.xxx.234:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKnIdb7zUWaGpK2QfG;rport
Contact: <sip:pre-ast-398-01 at 10.0.254.148:60099;transport=tls>
Max-Forwards: 70
From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.7
Content-Type: application/sdp
To: <sip:777 at sip.present.ca>;tag=as373d3af7
Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
CSeq: 3 INVITE
Content-Length: 698
v=0
o=- 60766 15292 IN IP4 10.0.254.148
s=bfymovv
c=IN IP4 10.0.254.148
t=0 0
m=audio 20902 RTP/SAVP 0 8 9 102 18 3 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:102 ILBC/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:f3zQ/60aXwqIKPDEQOPMlG/h6hEINZvMbCDs2EjW
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:f3zQ/60aXwqIKPDEQOPMlG/h6hEINZvMbCDs2EjW
a=fmtp:102 mode=20
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 57554 RTP/SAVP 34
a=rtpmap:34 H263/90000
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:vhKCfk6p7XjSag/uQ+mXeKN9QPR1ecwsbo1PC0NE
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:vhKCfk6p7XjSag/uQ+mXeKN9QPR1ecwsbo1PC0NE
a=fmtp:34 CIF=1;QCIF=2;SQCIF=2
a=sendrecv
<------------->
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 0 [ 56]: INVITE sip:777 at xxx.xxx.xxx.234:5061;transport=TLS SIP/2.0
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 1 [ 72]: Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKnIdb7zUWaGpK2QfG;rport
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 2 [ 62]: Contact: <sip:pre-ast-398-01 at 10.0.254.148:60099;transport=tls>
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 4 [ 84]: From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 5 [ 55]: Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 6 [ 25]: Supported: replaces, path
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 7 [ 45]: User-Agent: Acrobits Softphone Business/2.4.7
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 9 [ 43]: To: <sip:777 at sip.present.ca>;tag=as373d3af7
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 10 [ 49]: Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 11 [ 14]: CSeq: 3 INVITE
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 12 [ 19]: Content-Length: 698
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 13 [ 0]:
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 0 [ 3]: v=0
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 1 [ 35]: o=- 60766 15292 IN IP4 10.0.254.148
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 2 [ 9]: s=bfymovv
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.0.254.148
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 4 [ 5]: t=0 0
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 5 [ 41]: m=audio 20902 RTP/SAVP 0 8 9 102 18 3 101
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 7 [ 22]: a=rtpmap:102 ILBC/8000
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 8 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:f3zQ/60aXwqIKPDEQOPMlG/h6hEINZvMbCDs2EjW
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 9 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:f3zQ/60aXwqIKPDEQOPMlG/h6hEINZvMbCDs2EjW
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 10 [ 18]: a=fmtp:102 mode=20
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 13 [ 10]: a=ptime:20
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 14 [ 10]: a=sendrecv
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 15 [ 25]: m=video 57554 RTP/SAVP 34
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 16 [ 22]: a=rtpmap:34 H263/90000
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 17 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:vhKCfk6p7XjSag/uQ+mXeKN9QPR1ecwsbo1PC0NE
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 18 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:vhKCfk6p7XjSag/uQ+mXeKN9QPR1ecwsbo1PC0NE
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 19 [ 30]: a=fmtp:34 CIF=1;QCIF=2;SQCIF=2
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Body 20 [ 10]: a=sendrecv
[Jan 8 09:11:15] VERBOSE[15971] chan_sip.c: --- (13 headers 21 lines) ---
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: = Looking for Call ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A (Checking From) --From tag B7F2FA1CDE83A2DB7151A3C178C0BB26 --To-tag as373d3af7
[Jan 8 09:11:15] DEBUG[15971][C-00000003] logger.c: CALL_ID [C-00000003] bound to thread.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jan 8 09:11:15] DEBUG[15971][C-00000003] netsock2.c: Splitting '10.0.254.148:60099' into...
[Jan 8 09:11:15] DEBUG[15971][C-00000003] netsock2.c: ...host '10.0.254.148' and port '60099'.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: NAT detected for 10.0.254.148:60099 / xxx.xxx.xxx.52:60099
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Sending to xxx.xxx.xxx.52:60099 (NAT)
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing session-level SDP o=- 60766 15292 IN IP4 10.0.254.148... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing session-level SDP s=bfymovv... UNSUPPORTED OR FAILED.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] netsock2.c: Splitting '10.0.254.148' into...
[Jan 8 09:11:15] DEBUG[15971][C-00000003] netsock2.c: ...host '10.0.254.148' and port ''.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.254.148... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP audio format 0
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 0 based on m type on 0xb60b7988
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP audio format 8
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 8 based on m type on 0xb60b7988
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP audio format 9
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 9 based on m type on 0xb60b7988
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP audio format 102
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 102 based on m type on 0xb60b7988
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP audio format 18
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 18 based on m type on 0xb60b7988
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP audio format 3
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 3 based on m type on 0xb60b7988
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP audio format 101
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 101 based on m type on 0xb60b7988
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 101
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found audio description format ILBC for ID 102
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 ILBC/8000... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] sip/sdp_crypto.c: SRTP remote key unchanged; maintaining current policy
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:f3zQ/60aXwqIKPDEQOPMlG/h6hEINZvMbCDs2EjW... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:f3zQ/60aXwqIKPDEQOPMlG/h6hEINZvMbCDs2EjW... UNSUPPORTED OR FAILED.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:102 mode=20... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for ulaw to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for gsm to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for g723 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for adpcm to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for adpcm to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for lpc10 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for alaw to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for g722 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for slin to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for slin to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for adpcm to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for adpcm to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for g729 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for jpeg to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for h261 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for h263 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for silk8 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for ilbc to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for h263p to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for h264 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for silk12 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for ilbc to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for h263p to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for mpeg4 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for red to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for t140 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for silk16 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for silk24 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for speex to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for g726 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for g726aal2 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for siren14 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for g719 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for speex16 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for slin16 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting framing for speex32 to 20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found RTP video format 34
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Setting payload 34 based on m type on 0xb60b3644
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Found video description format H263 for ID 34
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (video) SDP a=rtpmap:34 H263/90000... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] sip/sdp_crypto.c: local_key64 61X8RStZ62fyiFTqKJZ+Le2I4WHCg6kzi1LgigFt len 40
[Jan 8 09:11:15] DEBUG[15971][C-00000003] res_srtp.c: Adding new policy for SSRC 1041645650
[Jan 8 09:11:15] DEBUG[15971][C-00000003] sip/sdp_crypto.c: SRTP policy activated
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (video) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:vhKCfk6p7XjSag/uQ+mXeKN9QPR1ecwsbo1PC0NE... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (video) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:vhKCfk6p7XjSag/uQ+mXeKN9QPR1ecwsbo1PC0NE... UNSUPPORTED OR FAILED.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (video) SDP a=fmtp:34 CIF=1;QCIF=2;SQCIF=2... OK.
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED.
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Capabilities: us - (gsm|ulaw|h263), peer - audio=(gsm|ulaw|alaw|g729|ilbc|g722)/video=(h263)/text=(nothing), combined - (gsm|ulaw|h263)
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jan 8 09:11:15] DEBUG[15971][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb737c2ec'
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Peer audio RTP is at port 10.0.254.148:20902
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 0 from 0xb60b7988 to 0xb737c498
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 3 from 0xb60b7988 to 0xb737c498
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 8 from 0xb60b7988 to 0xb737c498
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 9 from 0xb60b7988 to 0xb737c498
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 18 from 0xb60b7988 to 0xb737c498
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 101 from 0xb60b7988 to 0xb737c498
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 102 from 0xb60b7988 to 0xb737c498
[Jan 8 09:11:15] DEBUG[15971][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb737c2ec'
[Jan 8 09:11:15] DEBUG[15971][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb734aa7c'
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Peer video RTP is at port 10.0.254.148:57554
[Jan 8 09:11:15] DEBUG[15971][C-00000003] rtp_engine.c: Copying payload 34 from 0xb60b3644 to 0xb734ac28
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: We're settling with these formats: (gsm|ulaw|h263)
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Setting native formats after processing SDP. peer joint formats (gsm|ulaw|h263), old nativeformats (ulaw)
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Got a SIP re-invite for call DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: SIP/pre-ast-398-01-00000003: This call is UP....
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c:
<--- Transmitting (NAT) to xxx.xxx.xxx.52:60099 --->
SIP/2.0 100 Trying^M
Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKnIdb7zUWaGpK2QfG;received=xxx.xxx.xxx.52;rport=60099^M
From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26^M
To: <sip:777 at sip.present.ca>;tag=as373d3af7^M
Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A^M
CSeq: 3 INVITE^M
Server: PRE-IPT-SVC^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: <sip:777 at xxx.xxx.xxx.234:5061;transport=TLS>^M
Content-Length: 0^M
^M
<------------>
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKnIdb7zUWaGpK2QfG;received=xxx.xxx.xxx.52;rport=60099
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 2 [ 84]: From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 3 [ 43]: To: <sip:777 at sip.present.ca>;tag=as373d3af7
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 4 [ 49]: Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 6 [ 19]: Server: PRE-IPT-SVC
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 9 [ 52]: Contact: <sip:777 at xxx.xxx.xxx.234:5061;transport=TLS>
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 11 [ 0]:
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 100' onto TLS socket destined for xxx.xxx.xxx.52:60099
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: This call needs video offers!
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: ** Our capability: (gsm|ulaw|h263) Video flag: False Text flag: True
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: ** Our prefcodec: (nothing)
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Audio is at 12146
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Video is at xxx.xxx.xxx.234:12280
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Adding video codec 200002 (h263) to SDP
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: -- Done with adding codecs to SDP
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|h263)
[Jan 8 09:11:15] VERBOSE[15971][C-00000003] chan_sip.c:
<--- Reliably Transmitting (NAT) to xxx.xxx.xxx.52:60099 --->
SIP/2.0 200 OK^M
Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKnIdb7zUWaGpK2QfG;received=xxx.xxx.xxx.52;rport=60099^M
From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26^M
To: <sip:777 at sip.present.ca>;tag=as373d3af7^M
Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A^M
CSeq: 3 INVITE^M
Server: PRE-IPT-SVC^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Contact: <sip:777 at xxx.xxx.xxx.234:5061;transport=TLS>^M
Content-Type: application/sdp^M
Content-Length: 598^M
^M
v=0^M
o=root 2143900336 2143900337 IN IP4 xxx.xxx.xxx.234^M
s=Asterisk PBX 11.1.1^M
c=IN IP4 xxx.xxx.xxx.234^M
b=CT:384^M
t=0 0^M
m=audio 12146 RTP/SAVP 0 3 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:3 GSM/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=silenceSupp:off - - - -^M
a=ptime:20^M
a=sendrecv^M
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:9R9ZwGZs2xlYOtntWZUBqt4wBWgA5rqvo9kP34YE^M
m=video 12280 RTP/AVP 34^M
a=rtpmap:34 H263/90000^M
a=fmtp:34 SQCIF=2;QCIF=2;CIF=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0^M
a=sendrecv^M
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:61X8RStZ62fyiFTqKJZ+Le2I4WHCg6kzi1LgigFt^M
<------------>
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKnIdb7zUWaGpK2QfG;received=xxx.xxx.xxx.52;rport=60099
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 2 [ 84]: From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 3 [ 43]: To: <sip:777 at sip.present.ca>;tag=as373d3af7
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 4 [ 49]: Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 6 [ 19]: Server: PRE-IPT-SVC
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 9 [ 52]: Contact: <sip:777 at xxx.xxx.xxx.234:5061;transport=TLS>
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 11 [ 19]: Content-Length: 598
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Header 12 [ 0]:
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 0 [ 3]: v=0
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 1 [ 50]: o=root 2143900336 2143900337 IN IP4 xxx.xxx.xxx.234
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 2 [ 21]: s=Asterisk PBX 11.1.1
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 3 [ 23]: c=IN IP4 xxx.xxx.xxx.234
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 4 [ 8]: b=CT:384
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 5 [ 5]: t=0 0
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 6 [ 30]: m=audio 12146 RTP/SAVP 0 3 101
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - -
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 12 [ 10]: a=ptime:20
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 13 [ 10]: a=sendrecv
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 14 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:9R9ZwGZs2xlYOtntWZUBqt4wBWgA5rqvo9kP34YE
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 15 [ 24]: m=video 12280 RTP/AVP 34
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 16 [ 22]: a=rtpmap:34 H263/90000
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 17 [ 66]: a=fmtp:34 SQCIF=2;QCIF=2;CIF=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 18 [ 10]: a=sendrecv
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Body 19 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:61X8RStZ62fyiFTqKJZ+Le2I4WHCg6kzi1LgigFt
[Jan 8 09:11:15] DEBUG[15971][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for xxx.xxx.xxx.52:60099
[Jan 8 09:11:15] DEBUG[15971][C-00000003] logger.c: Call_ID [C-00000003] being removed from thread.
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- start learning mode pass with addr = xxx.xxx.xxx.52:20902
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- probation = 4, seq = 36386
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- Condition for learning hasn't exited, so reject the frame.
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- start learning mode pass with addr = xxx.xxx.xxx.52:20902
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- probation = 3, seq = 36387
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- Condition for learning hasn't exited, so reject the frame.
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- start learning mode pass with addr = xxx.xxx.xxx.52:20902
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- probation = 2, seq = 36388
[Jan 8 09:11:15] DEBUG[19238][C-00000003] res_rtp_asterisk.c: 0xb7316f98 -- Condition for learning hasn't exited, so reject the frame.
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 0 [ 53]: ACK sip:777 at xxx.xxx.xxx.234:5061;transport=TLS SIP/2.0
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 1 [ 72]: Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKp68TKXaEsc85IV1H;rport
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 3 [ 43]: To: <sip:777 at sip.present.ca>;tag=as373d3af7
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 4 [ 84]: From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 5 [ 49]: Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 6 [ 11]: CSeq: 3 ACK
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 7 [ 17]: Content-Length: 0
[Jan 8 09:11:15] DEBUG[15971] chan_sip.c: Header 8 [ 0]:
[Jan 8 09:11:15] VERBOSE[15971] chan_sip.c:
<--- SIP read from TLS:xxx.xxx.xxx.52:60099 --->
ACK sip:777 at xxx.xxx.xxx.234:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.0.254.148:60099;branch=z9hG4bKp68TKXaEsc85IV1H;rport
Max-Forwards: 70
To: <sip:777 at sip.present.ca>;tag=as373d3af7
From: "398" <sip:pre-ast-398-01 at sip.present.ca>;tag=B7F2FA1CDE83A2DB7151A3C178C0BB26
Call-ID: DFA7A5837DEA7A420C34A6EE380C4A9B107C301A
CSeq: 3 ACK
Content-Length: 0
)
> Video returned in SIP SDP RTP/AVP when it should be RTP/SAVP when SRTP (encryption=yes) is enabled.
> ---------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-20905
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20905
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/SRTP
> Affects Versions: 11.1.1
> Environment: Linux 2.6.32-279.19.1.el6.i686 #1 SMP Wed Dec 19 04:30:58 UTC 2012 i686 i686 i386 GNU/Linux
> Reporter: kris2k
> Severity: Minor
> Attachments: log.txt
>
>
> In a context where the SIP endpoint enforces the use of SRTP via SIP TLS, we noticed that the requested video was RTP/SAVP, when Asterisk returned a video feed being RTP/AVP.
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