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Problems in Grandstream Asterisk SRTP connection

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hemanshurpatel

Joined: 25 Jul 2009
Posts: 2

Posted: Sat Feb 27, 2010 7:08 am Post subject: Problems in Grandstream Asterisk SRTP connection

I am using asterisk with srtp support downloaded from http://svn.digium.com/svn/asterisk/team/group/srtp asterisk-srtp

i also install lib srtp
Compilation went ok, and i was been able to call when i am not using SRTP.

then i enabled srtp in my two grandstream device.
from them when i try to call to another GS device it gives me 488 not accepted error/

check the log:

Quote:

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:172.16.16.106:5060 --->
INVITE sip:1010@172.16.16.218:5090;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.16.106:5060;branch=z9hG4bK83f5a1b7d6823eb4
From: "1001" <sip:1001@172.16.16.218:5090;user=phone>;tag=f693c29592a1d421
To: <sip:1010@172.16.16.218:5090;user=phone>
Contact: <sip:1001@172.16.16.106:5060;user=phone>
Supported: replaces, timer, path
Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1010@172.16.16.218:5090;user=phone", nonce="2e16ca00", response="a09bae68ef9fbee1398127983156466b"
Call-ID: 36022fe491f56447@172.16.16.106
CSeq: 32546 INVITE
User-Agent: Grandstream GXV3000 1.1.3.29
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 953

v=0
o=1001 8000 8001 IN IP4 172.16.16.106
s=SIP Call
c=IN IP4 172.16.16.106
t=0 0
m=audio 5004 RTP/SAVP 18 0 8 4 2 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:avnP2xk9LAh9mk1kokQKow9OeRZGJmonRxXr3wvB
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:3Npw8Mo4lns0JUax+9IYZ8scL9+eCbpJOlB2P7HQ
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 5006 RTP/SAVP 99 103 34
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:w2ILrPlvubVM4wOUujXVYPz6lgsGYKjiirXrEHTf
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:KEbeFSVS7NnMQNF+CFvNl9Ha7V+qXo0+I3fPSVYe
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Eci==
a=rtpmap:103 H263-1998/90000
a=fmtp:103 QCIF=1 MaxBR=10240
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1 MaxBR=10240
a=framerate:30

<------------->
--- (14 headers 28 lines) ---
Sending to 172.16.16.106 : 5060 (no NAT)
Using INVITE request as basis request - 36022fe491f56447@172.16.16.106
Found peer '1001' for '1001' from 172.16.16.106:5060
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 101
Found RTP video format 99
Found RTP video format 103
Found RTP video format 34
Peer audio RTP is at port 172.16.16.106:5004
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Found video description format H264 for ID 99
Found video description format H263-1998 for ID 103
Found video description format H263 for ID 34
Capabilities: us - 0x28010c (ulaw|alaw|g729|h263|h264), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x380000 (h263|h263p|h264)/text=0x0 (nothing), combined - 0x28010c (ulaw|alaw|g729|h263|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.16.106:5004
Peer video RTP is at port 172.16.16.106:5006
Looking for 1010 in hemu_default (domain 172.16.16.218)
list_route: hop: <sip:1001@172.16.16.106:5060;user=phone>

<--- Transmitting (no NAT) to 172.16.16.106:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.16.106:5060;branch=z9hG4bK83f5a1b7d6823eb4;received=172.16.16.106
From: "1001" <sip:1001@172.16.16.218:5090;user=phone>;tag=f693c29592a1d421
To: <sip:1010@172.16.16.218:5090;user=phone>
Call-ID: 36022fe491f56447@172.16.16.106
CSeq: 32546 INVITE
Server: PANTHER-SC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:1010@172.16.16.218:5090>
Content-Length: 0

<------------>
-- Executing [1010@hemu_default:1] Set("SIP/1001-01e51ac8", "_SIPSRTP=optional") in new stack
-- Executing [1010@hemu_default:2] Set("SIP/1001-01e51ac8", "_SIPSRTP_CRYPTO=enable") in new stack
-- Executing [1010@hemu_default:3] Set("SIP/1001-01e51ac8", "_SIPSRTP=256") in new stack
-- Executing [1010@hemu_default:4] Dial("SIP/1001-01e51ac8", "SIP/1010,20,tT") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
Audio is at 172.16.16.218 port 17430
Video is at 172.16.16.218 port 17092
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.16.222:5064:
INVITE sip:1010@172.16.16.222:5064;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.16.218:5090;branch=z9hG4bK4c002f6e;rport
Max-Forwards: 70
From: "1001" <sip:1001@172.16.16.218:5090>;tag=as4ba3d2e3
To: <sip:1010@172.16.16.222:5064;user=phone>
Contact: <sip:1001@172.16.16.218:5090>
Call-ID: 6cfaa200351666ce3854a22006d11d4f@172.16.16.218
CSeq: 102 INVITE
User-Agent: PANTHER-SC
Date: Sat, 27 Feb 2010 06:54:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 500

v=0
o=root 1618678554 1618678554 IN IP4 172.16.16.218
s=Asterisk PBX UNKNOWN__and_probably_unsupported
c=IN IP4 172.16.16.218
b=CT:384
t=0 0
m=audio 17430 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 17092 RTP/SAVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MeTqYFkeFPA2oar4ThGezWIFHS88RO8vcdR/hEzY

---
-- Called 1010

<--- SIP read from UDP:172.16.16.222:5064 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.16.218:5090;branch=z9hG4bK4c002f6e;rport=5090
From: "1001" <sip:1001@172.16.16.218:5090>;tag=as4ba3d2e3
To: <sip:1010@172.16.16.222:5064;user=phone>
Call-ID: 6cfaa200351666ce3854a22006d11d4f@172.16.16.218
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXV3140 1.0.3.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.16.16.222:5064 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.16.218:5090;branch=z9hG4bK4c002f6e;rport=5090
From: "1001" <sip:1001@172.16.16.218:5090>;tag=as4ba3d2e3
To: <sip:1010@172.16.16.222:5064;user=phone>;tag=1653610119
Call-ID: 6cfaa200351666ce3854a22006d11d4f@172.16.16.218
CSeq: 102 INVITE
Contact: <sip:1010@172.16.16.222:5064;user=phone>
Supported: replaces, path, timer
User-Agent: Grandstream GXV3140 1.0.3.10
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
-- SIP/1010-01e62cd8 is ringing

<--- Transmitting (no NAT) to 172.16.16.106:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.16.106:5060;branch=z9hG4bK83f5a1b7d6823eb4;received=172.16.16.106
From: "1001" <sip:1001@172.16.16.218:5090;user=phone>;tag=f693c29592a1d421
To: <sip:1010@172.16.16.218:5090;user=phone>;tag=as026c6ec6
Call-ID: 36022fe491f56447@172.16.16.106
CSeq: 32546 INVITE
Server: PANTHER-SC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:1010@172.16.16.218:5090>
Content-Length: 0

<------------>

<--- SIP read from UDP:172.16.16.222:5064 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.16.16.218:5090;branch=z9hG4bK4c002f6e;rport=5090
From: "1001" <sip:1001@172.16.16.218:5090>;tag=as4ba3d2e3
To: <sip:1010@172.16.16.222:5064;user=phone>;tag=1653610119
Call-ID: 6cfaa200351666ce3854a22006d11d4f@172.16.16.218
CSeq: 102 INVITE
Contact: <sip:1010@172.16.16.222:5064;user=phone>
Supported: replaces, path, timer
User-Agent: Grandstream GXV3140 1.0.3.10
Warning: 304 GS "Media type not available"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Transmitting (no NAT) to 172.16.16.222:5064:
ACK sip:1010@172.16.16.222:5064;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.16.218:5090;branch=z9hG4bK4c002f6e;rport
Max-Forwards: 70
From: "1001" <sip:1001@172.16.16.218:5090>;tag=as4ba3d2e3
To: <sip:1010@172.16.16.222:5064;user=phone>;tag=1653610119
Contact: <sip:1001@172.16.16.218:5090>
Call-ID: 6cfaa200351666ce3854a22006d11d4f@172.16.16.218
CSeq: 102 ACK
User-Agent: PANTHER-SC
Content-Length: 0

---
-- SIP/1010-01e62cd8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [1010@hemu_default:5] Hangup("SIP/1001-01e51ac8", "") in new stack
== Spawn extension (hemu_default, 1010, 5) exited non-zero on 'SIP/1001-01e51ac8'
Scheduling destruction of SIP dialog '36022fe491f56447@172.16.16.106' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.16.16.106:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.16.16.106:5060;branch=z9hG4bK83f5a1b7d6823eb4;received=172.16.16.106
From: "1001" <sip:1001@172.16.16.218:5090;user=phone>;tag=f693c29592a1d421
To: <sip:1010@172.16.16.218:5090;user=phone>;tag=as026c6ec6
Call-ID: 36022fe491f56447@172.16.16.106
CSeq: 32546 INVITE
Server: PANTHER-SC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>

<--- SIP read from UDP:172.16.16.106:5060 --->
ACK sip:1010@172.16.16.218:5090;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.16.106:5060;branch=z9hG4bK83f5a1b7d6823eb4
From: "1001" <sip:1001@172.16.16.218:5090;user=phone>;tag=f693c29592a1d421
To: <sip:1010@172.16.16.218:5090;user=phone>;tag=as026c6ec6
Contact: <sip:1001@172.16.16.106:5060;user=phone>
Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1010@172.16.16.218:5090;user=phone", nonce="2e16ca00", response="a09bae68ef9fbee1398127983156466b"
Call-ID: 36022fe491f56447@172.16.16.106
CSeq: 32546 ACK
User-Agent: Grandstream GXV3000 1.1.3.29
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '6cfaa200351666ce3854a22006d11d4f@172.16.16.218' Method: INVITE
Really destroying SIP dialog 'dd81fe60a520c210@172.16.16.106' Method: REGISTER

Note the invite request forwarded to another GS phone....it has not only one crypto tag in its SDP.
Can anyone suggest what is wrong here?

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