<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">just a quick observation, but not sure that it is critical<div><br></div><div>in this case, the first invite comes without Authorization header, then gets challenged then resends the invite (with increased cseq) with calculated response based on the challenge from the server.</div><div><br></div><div>In your AAstra case, the first invite already contained Authorization header (which is really impossible because you don't have all the pieces to calculate the response). Normally not an issue, as UAS should challenge it, but I wonder why it does it anyway. I would compare Authorize elements between 2 cases particularly response, uri and authorization user name. if response is the same between the two, I am lost.<br><div><div>On Nov 24, 2011, at 2:11 PM, Jonas Kellens wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">
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On 11/22/2011 06:13 PM, Alex Vishnev wrote:
<blockquote cite="mid:603CFA27-5247-4C07-95D6-E061B8412522@gmail.com" type="cite">
<pre wrap="">it is strange that Aastra acks 401, sends another invite but does not increase CSeq. Is that the same behavior with others?
On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote:
</pre>
</blockquote>
<font face="Helvetica, Arial, sans-serif">This is a trace taken when
an Alcatel-Lucent PBX sends an INVITE (no refusal by Asterisk). Do
you see any difference ?<br>
<br>
</font><font face="Helvetica, Arial, sans-serif">A1.A1.A1.A1 =
IP-address Asterisk PBX<br>
AL.AL.AL.AL = IP-address Alcatel-Lucent PBX<br>
<br>
<br>
<--- SIP read from UDP:AL.AL.AL.AL:5060 ---><br>
INVITE <a class="moz-txt-link-freetext" href="sip:311083335533@A1.A1.A1.A1;user=phone">sip:311083335533@A1.A1.A1.A1;user=phone</a> SIP/2.0<br>
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE,
OPTIONS, UPDATE<br>
Supported: replaces, timer, 100rel<br>
User-Agent: OmniPCX Enterprise R9.1 i1.605.21<br>
Session-Expires: 1800;refresher=uac<br>
Min-SE: 900<br>
P-Asserted-Identity: "Dan Luc"
<a class="moz-txt-link-rfc2396E" href="sip:328883300@192.168.8.10;user=phone"><sip:328883300@192.168.8.10;user=phone></a><br>
To: <a class="moz-txt-link-rfc2396E" href="sip:311083335533@A1.A1.A1.A1;user=phone"><sip:311083335533@A1.A1.A1.A1;user=phone></a><br>
From: "Dan Luc"
<a class="moz-txt-link-rfc2396E" href="sip:328883300@AL.AL.AL.AL:5060;user=phone"><sip:328883300@AL.AL.AL.AL:5060;user=phone></a>;tag=37a49f0486bab42b240be214b2d13153<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:328883300@AL.AL.AL.AL:5060;transport=UDP"><sip:328883300@AL.AL.AL.AL:5060;transport=UDP></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:2fae0b0266919172cac1e23dc2567cd2@192.168.8.10">2fae0b0266919172cac1e23dc2567cd2@192.168.8.10</a><br>
CSeq: 443337258 INVITE<br>
Via: SIP/2.0/UDP
AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae<br>
Max-Forwards: 70<br>
Content-Type: application/sdp<br>
Content-Length: 292<br>
<br>
v=0<br>
o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL<br>
s=abs<br>
c=IN IP4 AL.AL.AL.AL<br>
t=0 0<br>
m=audio 34422 RTP/AVP 8 18 97<br>
a=sendrecv<br>
a=rtpmap:8 PCMA/8000<br>
a=ptime:20<br>
a=maxptime:30<br>
a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>
a=ptime:20<br>
a=maxptime:40<br>
a=rtpmap:97 telephone-event/8000<br>
<br>
<br>
<--- Reliably Transmitting (NAT) to AL.AL.AL.AL:5060 ---><br>
SIP/2.0 401 Unauthorized<br>
Via: SIP/2.0/UDP
AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae;received=AL.AL.AL.AL<br>
From: "Dan Luc"
<a class="moz-txt-link-rfc2396E" href="sip:328883300@AL.AL.AL.AL:5060;user=phone"><sip:328883300@AL.AL.AL.AL:5060;user=phone></a>;tag=37a49f0486bab42b240be214b2d13153<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:311083335533@A1.A1.A1.A1;user=phone"><sip:311083335533@A1.A1.A1.A1;user=phone></a>;tag=as1b6f387a<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:2fae0b0266919172cac1e23dc2567cd2@192.168.8.10">2fae0b0266919172cac1e23dc2567cd2@192.168.8.10</a><br>
CSeq: 443337258 INVITE<br>
Server: Asterisk PBX 1.6.2.20<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO<br>
Supported: replaces, timer<br>
WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld",
nonce="7684ab1d"<br>
Content-Length: 0<br>
<br>
<br>
<--- SIP read from UDP:AL.AL.AL.AL:5060 ---><br>
INVITE <a class="moz-txt-link-freetext" href="sip:311083335533@A1.A1.A1.A1;user=phone">sip:311083335533@A1.A1.A1.A1;user=phone</a> SIP/2.0<br>
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE,
OPTIONS, UPDATE<br>
Supported: replaces, timer, 100rel<br>
User-Agent: OmniPCX Enterprise R9.1 i1.605.21<br>
Session-Expires: 1800;refresher=uac<br>
Min-SE: 900<br>
P-Asserted-Identity: "Dan Luc"
<a class="moz-txt-link-rfc2396E" href="sip:328883300@192.168.8.10;user=phone"><sip:328883300@192.168.8.10;user=phone></a><br>
To: <a class="moz-txt-link-rfc2396E" href="sip:311083335533@A1.A1.A1.A1;user=phone"><sip:311083335533@A1.A1.A1.A1;user=phone></a><br>
From: "Dan Luc"
<a class="moz-txt-link-rfc2396E" href="sip:328883300@AL.AL.AL.AL:5060;user=phone"><sip:328883300@AL.AL.AL.AL:5060;user=phone></a>;tag=37a49f0486bab42b240be214b2d13153<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:328883300@AL.AL.AL.AL:5060;transport=UDP"><sip:328883300@AL.AL.AL.AL:5060;transport=UDP></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:2fae0b0266919172cac1e23dc2567cd2@192.168.8.10">2fae0b0266919172cac1e23dc2567cd2@192.168.8.10</a><br>
CSeq: 443337259 INVITE<br>
Max-Forwards: 70<br>
Authorization: Digest
username="SIPPEERusername",realm="domain.tld",nonce="7684ab1d",algorithm=MD5,uri=<a class="moz-txt-link-rfc2396E" href="sip:311083335533@A1.A1.A1.A1;user=phone">"sip:311083335533@A1.A1.A1.A1;user=phone"</a>,response="38bb824b9081bf2eefe9f9677d3eb005"<br>
Via: SIP/2.0/UDP
AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726<br>
Content-Type: application/sdp<br>
Content-Length: 292<br>
<br>
v=0<br>
o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL<br>
s=abs<br>
c=IN IP4 AL.AL.AL.AL<br>
t=0 0<br>
m=audio 34422 RTP/AVP 8 18 97<br>
a=sendrecv<br>
a=rtpmap:8 PCMA/8000<br>
a=ptime:20<br>
a=maxptime:30<br>
a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>
a=ptime:20<br>
a=maxptime:40<br>
a=rtpmap:97 telephone-event/8000<br>
<br>
<br>
<--- Transmitting (NAT) to AL.AL.AL.AL:5060 ---><br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP
AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726;received=AL.AL.AL.AL<br>
From: "Dan Luc"
<a class="moz-txt-link-rfc2396E" href="sip:328883300@AL.AL.AL.AL:5060;user=phone"><sip:328883300@AL.AL.AL.AL:5060;user=phone></a>;tag=37a49f0486bab42b240be214b2d13153<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:311083335533@A1.A1.A1.A1;user=phone"><sip:311083335533@A1.A1.A1.A1;user=phone></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:2fae0b0266919172cac1e23dc2567cd2@192.168.8.10">2fae0b0266919172cac1e23dc2567cd2@192.168.8.10</a><br>
CSeq: 443337259 INVITE<br>
Server: Asterisk PBX 1.6.2.20<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO<br>
Supported: replaces, timer<br>
Session-Expires: 1800;refresher=uac<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:311083335533@A1.A1.A1.A1"><sip:311083335533@A1.A1.A1.A1></a><br>
Content-Length: 0</font><br>
<br>
<br>
Thanks !<br>
<br>
Jonas.<br>
</div>
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