<div>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.</div>
<div> </div>
<div>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.
</div>
<div> </div>
<div>Asterisk sends 200 OK:</div>
<div>o=root 16300 16300 IN IP4 203.89.nnn.nnn</div>
<div> </div>
<div>
<div>Asterisk sends 200 OK (retransmission):</div></div>
<div>o=root 16300 16301 IN IP4 203.89.nnn.nnn</div>
<div> </div>
<div>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?
</div>
<div> </div>
<div>Raj</div>
<div> </div>
<div> </div>
<div> </div>
<div> </div>
<div><span class="gmail_quote">On 4/1/07, <b class="gmail_sendername">kjcsb</b> <<a href="mailto:kjcsb@yahoo.com">kjcsb@yahoo.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">>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.
<br><br>>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.
<br><br>OK. But in the calls that don't get dropped, the "branch=0" is present also. See below for an example:<br><br><-- SIP read from 147.202.nnn.nnn:5060:<br>INVITE <a href="mailto:sip:6499777777@203.89.nnn.nnn">
sip:6499777777@203.89.nnn.nnn</a> SIP/2.0<br>Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on><br>Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0<br>Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
<br>From: "6494444444" <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>>;tag=as1370b1ab<br>To: <<a href="mailto:sip:6499777777@domain.co.nz">sip:6499777777@domain.co.nz
</a>><br>Contact: <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>><br>Call-ID: <a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn
</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 69<br>Date: Mon, 02 Apr 2007 03:37:54 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Content-Type: application/sdp<br>Content-Length: 338
<br>v=0<br>o=root 11402 11402 IN IP4 202.180.nnn.nnn<br>s=session<br>c=IN IP4 202.180.nnn.nnn<br>t=0 0<br>m=audio 39686 RTP/AVP 18 97 3 0 8 101<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:97 iLBC/8000<br>a=rtpmap:3 GSM/8000
<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>--- (15 headers 15 lines) ---<br>Using INVITE request as basis request - <a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">
1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn</a><br>Sending to 147.202.nnn.nnn : 5060 (non-NAT)<br>Found peer 'DLS'<br>Found RTP audio format 18<br>Found RTP audio format 97<br>Found RTP audio format 3<br>Found RTP audio format 0
<br>Found RTP audio format 8<br>Found RTP audio format 101<br>Peer audio RTP is at port 202.180.nnn.nnn:39686<br>Found description format G729<br>Found description format iLBC<br>Found description format GSM<br>Found description format PCMU
<br>Found description format PCMA<br>Found description format telephone-event<br>Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)<br>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
<br>Looking for 6499777777 in from-trunk (domain 203.89.nnn.nnn)<br>list_route: hop: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on><br>Transmitting (no NAT) to 147.202.nnn.nnn:5060:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP
147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn<br>Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060<br>From: "6494444444" <<a href="mailto:sip:6494444444@202.180.nnn.nnn">
sip:6494444444@202.180.nnn.nnn</a>>;tag=as1370b1ab<br>To: <<a href="mailto:sip:6499777777@domain.co.nz">sip:6499777777@domain.co.nz</a>><br>Call-ID: <a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">
1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip:6499777777@203.89.nnn.nnn">
sip:6499777777@203.89.nnn.nnn</a>><br>Content-Length: 0<br><br>---<br> -- Goto (ivr-3,s,1)<br> -- Executing Set("SIP/6499777777-b7908550", "LOOPCOUNT=0") in new stack<br> -- Executing Set("SIP/6499777777-b7908550", "__DIR-CONTEXT=11000111000") in new stack
<br> -- Executing Answer("SIP/6499777777-b7908550", "") in new stack<br>We're at 203.89.nnn.nnn port 15804<br>Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x8 (alaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP
<br>Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn<br>Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
<br>Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on><br>From: "6494444444" <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>>;tag=as1370b1ab<br>To: <<a href="mailto:sip:6499777777@domain.co.nz">
sip:6499777777@domain.co.nz</a>>;tag=as7ecf44d1<br>Call-ID: <a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX
<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip:6499777777@203.89.nnn.nnn">sip:6499777777@203.89.nnn.nnn</a>><br>Content-Type: application/sdp<br>Content-Length: 244
<br>v=0<br>o=root 16300 16300 IN IP4 203.89.nnn.nnn<br>s=session<br>c=IN IP4 203.89.nnn.nnn<br>t=0 0<br>m=audio 15804 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>---<br> -- Executing Wait("SIP/6499777777-b7908550", "1") in new stack<br>capetown*CLI><br><-- SIP read from 147.202.nnn.nnn:5060:<br>ACK <a href="mailto:sip:6499777777@203.89.nnn.nnn">
sip:6499777777@203.89.nnn.nnn</a> SIP/2.0<br>Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on><br>Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0<br>Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK5ba4f251;rport=5060
<br>From: "6494444444" <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>>;tag=as1370b1ab<br>To: <<a href="mailto:sip:6499777777@domain.co.nz">sip:6499777777@domain.co.nz
</a>>;tag=as7ecf44d1<br>Contact: <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>><br>Call-ID: <a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn
</a><br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX<br>Max-Forwards: 69<br>Content-Length: 0<br><br>--- (12 headers 0 lines) ---<br> -- Executing Set("SIP/6499777777-b7908550", "TIMEOUT(digit)=3") in new stack
<br> -- Digit timeout set to 3<br> -- Executing Set("SIP/6499777777-b7908550", "TIMEOUT(response)=10") in new stack<br> -- Response timeout set to 10<br> -- Executing BackGround("SIP/6499777777-b7908550", "custom/11000111000-welcome") in new stack
<br> -- Playing 'custom/11000111000-welcome' (language 'nz')<br>capetown*CLI><br><-- SIP read from 147.202.nnn.nnn:5060:<br>BYE <a href="mailto:sip:6499777777@203.89.nnn.nnn">sip:6499777777@203.89.nnn.nnn
</a> SIP/2.0<br>Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on><br>Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK32ab.feee6b67.0<br>Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK7916f637;rport=5060<br>
From: "6494444444" <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>>;tag=as1370b1ab<br>To: <<a href="mailto:sip:6499777777@domain.co.nz">sip:6499777777@domain.co.nz</a>
>;tag=as7ecf44d1<br>Contact: <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>><br>Call-ID: <a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn
</a><br>CSeq: 103 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 69<br>Content-Length: 0<br><br>--- (12 headers 0 lines) ---<br>Sending to 147.202.nnn.nnn : 5060 (non-NAT)<br>Transmitting (no NAT) to 147.202.nnn.nnn:5060:
<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK32ab.feee6b67.0;received=147.202.nnn.nnn<br>Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK7916f637;rport=5060<br>Record-Route: <sip:147.202.nnn.nnn
;ftag=as1370b1ab;lr=on><br>From: "6494444444" <<a href="mailto:sip:6494444444@202.180.nnn.nnn">sip:6494444444@202.180.nnn.nnn</a>>;tag=as1370b1ab<br>To: <<a href="mailto:sip:6499777777@domain.co.nz">
sip:6499777777@domain.co.nz</a>>;tag=as7ecf44d1<br>Call-ID: <a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn</a><br>CSeq: 103 BYE<br>User-Agent: Asterisk PBX
<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip:6499777777@203.89.nnn.nnn">sip:6499777777@203.89.nnn.nnn</a>><br>Content-Length: 0<br><br>---<br>== Spawn extension (ivr-3, s, 7) exited non-zero on 'SIP/6499777777-b7908550'
<br> -- Executing Hangup("SIP/6499777777-b7908550", "") in new stack<br>== Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/6499777777-b7908550'<br>Destroying call '<a href="mailto:1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn">
1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn</a>'<br>capetown*CLI> sip no debug<br>SIP Debugging Disabled<br>capetown*CLI><br><br><br>Cameron<br><br><br><br><br><br>___________________________________________________________
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