I found a subtle difference between the two traces you sent (the call that
works and the call that gets dropped). This may or may not be what's causing
the problem.
The call that gets dropped had a retransmission of INVITE from UAC to
UAS (and therefore retransmission of 200 OK from UAS to UAC). There is
nothing wrong with the re-transmission as such, but I noticed a
potential bug in Asterisk in the way it responds to an
INVITE retransmission. Asterisk is bumping up the session version number in
the retransmitted 200 OK's SDP. This is as if Asterisk is treating the
INVITE retransmission as a RE-INVITE.
Asterisk sends 200 OK:
o=root 16300 16300 IN IP4 203.89.nnn.nnn
Asterisk sends 200 OK (retransmission):
o=root 16300 16301 IN IP4 203.89.nnn.nnn
Ideally, this bug should have nothing to do with why Asterisk is ignoring
the ACK (which is why it keeps reatrasmitting the 200 OK and eventually
drops the call). However, if you can confirm that all dropped calls have
INVITE retransmission then that might give us a clue?
Raj
On 4/1/07, kjcsb <[EMAIL PROTECTED]> wrote:
>One potential reason could be that the ACK request being sent to Asterisk
is malformed. Notice >"branch=0" in the top Via. This should start with
"z9hG4bK" magic cookie since the INVITE was an RFC >3261 transaction.
>While "branch=0" is valid in RFC 2543, I don't think an INVITE can
start-off as RFC 3261 and then the >ACK can switch over to RFC 2543 in the
middle of the transaction. Clearly, Asterisk is dropping this ACK >on the
floor.
OK. But in the calls that don't get dropped, the "branch=0" is present
also. See below for an example:
<-- SIP read from 147.202.nnn.nnn:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
From: "6494444444" <sip:[EMAIL PROTECTED]>;tag=as1370b1ab
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Mon, 02 Apr 2007 03:37:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 11402 11402 IN IP4 202.180.nnn.nnn
s=session
c=IN IP4 202.180.nnn.nnn
t=0 0
m=audio 39686 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 15 lines) ---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 147.202.nnn.nnn : 5060 (non-NAT)
Found peer 'DLS'
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 202.180.nnn.nnn:39686
Found description format G729
Found description format iLBC
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e
(gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 6499777777 in from-trunk (domain 203.89.nnn.nnn)
list_route: hop: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=
147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
From: "6494444444" <sip:[EMAIL PROTECTED]>;tag=as1370b1ab
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
-- Goto (ivr-3,s,1)
-- Executing Set("SIP/6499777777-b7908550", "LOOPCOUNT=0") in new stack
-- Executing Set("SIP/6499777777-b7908550",
"__DIR-CONTEXT=11000111000") in new stack
-- Executing Answer("SIP/6499777777-b7908550", "") in new stack
We're at 203.89.nnn.nnn port 15804
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=
147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
From: "6494444444" <sip:[EMAIL PROTECTED]>;tag=as1370b1ab
To: <sip:[EMAIL PROTECTED]>;tag=as7ecf44d1
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 16300 16300 IN IP4 203.89.nnn.nnn
s=session
c=IN IP4 203.89.nnn.nnn
t=0 0
m=audio 15804 RTP/AVP 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 - - - -
---
-- Executing Wait("SIP/6499777777-b7908550", "1") in new stack
capetown*CLI>
<-- SIP read from 147.202.nnn.nnn:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK5ba4f251;rport=5060
From: "6494444444" <sip:[EMAIL PROTECTED]>;tag=as1370b1ab
To: <sip:[EMAIL PROTECTED]>;tag=as7ecf44d1
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
--- (12 headers 0 lines) ---
-- Executing Set("SIP/6499777777-b7908550", "TIMEOUT(digit)=3") in new
stack
-- Digit timeout set to 3
-- Executing Set("SIP/6499777777-b7908550", "TIMEOUT(response)=10") in
new stack
-- Response timeout set to 10
-- Executing BackGround("SIP/6499777777-b7908550",
"custom/11000111000-welcome") in new stack
-- Playing 'custom/11000111000-welcome' (language 'nz')
capetown*CLI>
<-- SIP read from 147.202.nnn.nnn:5060:
BYE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK32ab.feee6b67.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK7916f637;rport=5060
From: "6494444444" <sip:[EMAIL PROTECTED]>;tag=as1370b1ab
To: <sip:[EMAIL PROTECTED]>;tag=as7ecf44d1
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
--- (12 headers 0 lines) ---
Sending to 147.202.nnn.nnn : 5060 (non-NAT)
Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK32ab.feee6b67.0;received=
147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK7916f637;rport=5060
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
From: "6494444444" <sip:[EMAIL PROTECTED]>;tag=as1370b1ab
To: <sip:[EMAIL PROTECTED]>;tag=as7ecf44d1
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
== Spawn extension (ivr-3, s, 7) exited non-zero on
'SIP/6499777777-b7908550'
-- Executing Hangup("SIP/6499777777-b7908550", "") in new stack
== Spawn extension (ivr-3, h, 1) exited non-zero on
'SIP/6499777777-b7908550'
Destroying call '[EMAIL PROTECTED]'
capetown*CLI> sip no debug
SIP Debugging Disabled
capetown*CLI>
Cameron
___________________________________________________________
New Yahoo! Mail is the ultimate force in competitive emailing. Find out
more at the Yahoo! Mail Championships. Plus: play games and win prizes.
http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
