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2 Replies Last post: Mar 1, 2010 11:36 PM by maxglucksmann  
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maxglucksmann
4 posts since
Mar 1, 2010

Mar 1, 2010 10:10 PM

ACK TIMEOUT - No response to OpenSBC from Asterisk


Good evening,


We're trying to use OpenSBC to proxy calls from an Asterisk server.

Our OpenSBC is registering with a SIP trunk and our Asterisk server registers in OpenSBC. We're not using Upper Registration.

When sending calls, the call is forwarded to the SIP trunk accordingly, but once the call is connected and OpenSBC sends the CSeq message 102 to the INVITE with a 200 OK, for some reason Asterisk doesn't send back an ACK message and OpenSBC hungs up giving a ACK TIMEOUT error.


This is one of the 200 OK messages in Asterisk's console:

pbx1*CLI>
<--- SIP read from xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 200 OK
From: "5555558888" <sip:5555555555@yyy.yyy.yyy.yyy>;tag=as1b56c719
To: <sip:555666777777@xxx.xxx.xxx.xxx>;tag=5ae8224f1424df11955bb1dafd175bf5
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK18a69deb;rport=5060;received=yyy.yyy.yyy.yyy
CSeq: 102 INVITE
Call-ID: 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy
Contact: <sip:555666777777@xxx.xxx.xxx.xxx:5060>
Record-Route: <sip:xxx.xxx.xxx.xxx:5060;lr>
Server: SIP Trunk
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 25214 25215 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 10016 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10018 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=sendrecv

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

No response is sent by Astyerisk. Meanwhile, the log of OpenSBC looks like this:

      • CALL ESTABLISHED ***
ACK sip:19543666429@xxx.xxx.xxx.xxx:5064 SIP/2.0 Method(ACK) DST: xxx.xxx.xxx.xxx:5064:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=483
<<< ACK sip:19543666429@xxx.xxx.xxx.xxx:5064 SIP/2.0 Method(ACK) SRC: xxx.xxx.xxx.xxx:5060:UDP enc=0 bytes=483
SIP/2.0 200 OK Method(INVITE) DST: yyy.yyy.yyy.yyy:5060:UDP src=xxx.xxx.xxx.xxx:5061 enc=0 bytes=917
200 Ok - 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy
SIP/2.0 200 OK Method(INVITE) DST: yyy.yyy.yyy.yyy:5060:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=907
200 Ok - 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy
SIP/2.0 200 OK Method(INVITE) DST: yyy.yyy.yyy.yyy:5060:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=907
200 Ok - 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy
SIP/2.0 200 OK Method(INVITE) DST: yyy.yyy.yyy.yyy:5060:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=907
RETRANSMIT: 200 Ok - 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy
SIP/2.0 200 OK Method(INVITE) DST: yyy.yyy.yyy.yyy:5060:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=907
RETRANSMIT: 200 Ok - 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy
SIP/2.0 200 OK Method(INVITE) DST: yyy.yyy.yyy.yyy:5060:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=907
RETRANSMIT: 200 Ok - 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy
SIP/2.0 200 OK Method(INVITE) DST: yyy.yyy.yyy.yyy:5060:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=907
! ACK Timeout !
      • B2BUACONNECTION DESTRUCTION SEQUENCE STARTED
      • CALL TEAR DOWN *** 4690dbc52e0f43d86e2396be1cd1f5f4@yyy.yyy.yyy.yyy-connection
BYE sip:5555558888@yyy.yyy.yyy.yyy SIP/2.0 Method(BYE) DST: yyy.yyy.yyy.yyy:5060:UDP SRC: xxx.xxx.xxx.xxx:5060 enc=0 bytes=487

Any guidance will be greatly appreciated.


Thanks,

Max Glucksmann

Comtel Networks

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joegen
519 posts since
Apr 28, 2007
1. Mar 1, 2010 11:20 PM in response to: maxglucksmann
Re: ACK TIMEOUT - No response to OpenSBC from Asterisk
I don't want to sound annoying but have you tried asking this question to the asterisk mailing list? The key in solving this is to pinpoint what causes asterisk to ignore the 200 OK. As far as I can tell, it's perfectly constructed.
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maxglucksmann
4 posts since
Mar 1, 2010
2. Mar 1, 2010 11:36 PM in response to: joegen
Re: ACK TIMEOUT - No response to OpenSBC from Asterisk

Thanks, I also thought it was an Asterisk issue, but I was wondering if anyone here had encountered it too.


Happily I tested several settings and was able to connect...


The key was to set in the peer settings:

canreinvite=yes


Hope this helps sombody else. Now the ACK was properly sent:

Found RTP audio format 0
Found RTP video format 31
Found RTP video format 34
[Mar 1 23:31:05] DEBUG[8628]: chan_sip.c:5422 process_sdp: Peer doesn't provide T.38 UDPTL
Peer audio RTP is at port xxx.xxx.xxx.xxx:10048
Found audio description format PCMU for ID 0
Found video description format H261 for ID 31
Found video description format H263 for ID 34
Capabilities: us - 0x3c0104 (ulaw|g729|h261|h263|h263p|h264), peer - audio=0xc0004 (ulaw|h261|h263)/video=0xc0000 (h261|h263), combined - 0xc0004 (ulaw|h261|h263)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port xxx.xxx.xxx.xxx:10048
Peer video RTP is at port xxx.xxx.xxx.xxx:10050
list_route: hop: <sip:xxx.xxx.xxx.xxx:5060;lr>
set_destination: Parsing <sip:xxx.xxx.xxx.xxx:5060;lr> for address/port to send to
set_destination: set destination to xxx.xxx.xxx.xxx, port 5060
Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
ACK sip:5556667777@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2723db44;rport
Route: <sip:xxx.xxx.xxx.xxx:5060;lr>
From: "5555558888" <sip:5555558888@yyy.yyy.yyy.yyy>;tag=as5c1caf40
To: <sip:5556667777 at sbc dot astratel dot ca>;tag=aa2d77522224df118e3cfb8bdc4f3cee
Contact: <sip:5555558888@yyy.yyy.yyy.yyy>
Call-ID: 43030e7333549e2671ed689219ee039f@yyy.yyy.yyy.yyy
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
-- SIP/usa_unlimited-05515c60 answered SIP/101-c0033090
-- fixed jitterbuffer created on channel SIP/usa_unlimited-05515c60
-- fixed jitterbuffer created on channel SIP/101-c0033090

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