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Jon Steer
2013-Jan-22 19:26 UTC
[asterisk-users] Audio not decrypted between Asterisk and encrypted client

I am trying to setup a PCMU conference call using either Blink Lite/and or
a webRTC client. I have SDES on for RTP but NOT SSL for the SIP Channel.
 The audio coming from asterisk is fine, but it would appear that the audio
going to Asterisk is not being decoded correctly and results in the
conference recording being a giant ball of noise.  Wireshark sees audio
traffic in both directions, however, the conference channel says it doesn't
receive any audio resulting in a recording with static.

There are two problems here, one , at the decode point of the incoming RTP
stream, apparently the audio decode on the incoming asterisk side is
failing silently, probably because of something to do with the decryption
suite selection. Secondly, the conference channel thinks there is incoming
audio, when there isn't and just records noise.

This happens on both Blink and webRTC.  I am more concerned with the webRTC
case.

Environment
 Asterisk SVN-trunk-r379495M built by root @xxxx on a x86_64 on Fedora 18
 Blink Lite on Mac OS/X
 webRTC on Chrome Canary Version 26.0.1390.0 canary


INVITE sip:conf at starwars.local SIP/2.0
Via: SIP/2.0/UDP 10.173.107.233:52263
;rport;branch=z9hG4bKPjAAeszOcFhHjIPFEYpJYhafDAYM.pEY4t
Max-Forwards: 70
From: "Govner Bad" <sip:guv at
starwars.local>;tag=CllM3sPM7CH15pQedTmgTytpuNi-UJZOTo: <sip:conf at starwars.local>
Contact: <sip:64295807 at 10.173.107.233:52263>
Call-ID: peGYqLeBpNHjtN8Af2.pVfwHEA3zX6XX
CSeq: 1280 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE,
REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink Lite 2.0.1 (MacOSX)
Authorization: Digest username="guv",
realm="starwars.local",
nonce="5c41c128", uri="sip:conf at starwars.local",
response="2a8636d66679731ceb858f077a1e821c", algorithm=MD5
Content-Type: application/sdp
Content-Length: 617

v=0
o=- 3567865409 3567865409 IN IP4 10.173.107.233
s=Blink Lite 2.0.1 (MacOSX)
c=IN IP4 10.173.107.233
t=0 0
m=audio 53946 RTP/SAVP 0 8 101
a=rtcp:59154
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:8C4kwuFbwUE/+ufBZBlmOCiwfJ3lreAJzmNMUSeM
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:9RLQwXBkzFlXUu9mpiXEcRSeASZA8/EV+Hrzr2KV
a=ice-ufrag:34d60893
a=ice-pwd:4fcd05fe
a=candidate:Haad6be9 1 UDP 2130706431 10.173.107.233 53946 typ host
a=candidate:Haad6be9 2 UDP 2130706430 10.173.107.233 59154 typ host
a

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.173.107.233:52263
;branch=z9hG4bKPjAAeszOcFhHjIPFEYpJYhafDAYM.pEY4t;received=10.173.107.233;rport=52263
From: "Govner Bad" <sip:guv at starwars
.local>;tag=CllM3sPM7CH15pQedTmgTytpuNi-UJZO
To: <sip:conf at starwars.local>;tag=as2e1a60c2
Call-ID: peGYqLeBpNHjtN8Af2.pVfwHEA3zX6XX
CSeq: 1280 INVITE
Server: BSLocal
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:conf at 10.173.107.240:5060>
Content-Type: application/sdp
Content-Length: 567

v=0
o=root 1094504443 1094504443 IN IP4 10.173.107.240
s=BSLocal
c=IN IP4 10.173.107.240
t=0 0
m=audio 18186 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:2a8bd4ba1e58c3847aa9d5a03d73d61a
a=ice-pwd:08c4a4c228aa018108b48ffe5a565e4a
a=candidate:Haad6bf0 1 UDP 2130706431 10.173.107.240 18186 typ host
a=candidate:Haad6bf0 2 UDP 2130706430 10.173.107.240 18187 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:2lSCjAPw1OeCb1w67J6FNHV0VxPO2F4yeh1CdPWd

[sip.conf]
nat=no
icesupport=yes
directmedia=yes
directmedia=nonat
encryption=yes
encryption_taglen=80
avpf=yes

[rtp.conf]
icesupport=yes

[users.conf]

[guv]
type=peer
username=guv
host=dynamic
secret=password
context=educateonline
hasiax = no
hassip = yes
encryption = yes
avpf = no
icesupport = yes
videosupport=no
directmedia=yes
;stun=stun.l.google.com:19302


transport=ws,udp
subscribemwi=no
hasvoicemail=no


[confbridge]
dsp_drop_silence=yes
denoise=yes
dtmf_passthrough=no



-- 
"Don't stand still, if you see me running down the road, 'cause
there is
trouble right behind me".
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asterisk users - Jan 2013 - Audio not decrypted between Asterisk and encrypted client

[asterisk-users] Audio not decrypted between Asterisk and encrypted client
