[Asterisk-Users] Fwd: Message from iptel.org SIP admin (more register= bugs)

John Todd jtodd at loligo.com
Mon Feb 24 12:40:23 MST 2003


Bug in the register= code; see details below from the developer of 
"ser" (SIP Express Router)

Apparently, ACKs don't need to be sent on OK's to REGISTERs.  Plus, 
malformed data somewhere... no details on that, though.

JT




>Date: Sun, 23 Feb 2003 23:54:07 +0100
>To: John Todd <jtodd at loligo.com>
>From: Jiri Kuthan <jiri at iptel.org>
>Subject: Re: Message from iptel.org SIP admin
>
>At 11:42 PM 2/23/2003, John Todd wrote:
>>While I understand SIP to a "reasonable" degree, I don't see what 
>>the problem is below with the >ACK message.  Should it not be 
>>sending an ACK at all after it receives an "OK"?   I've not 
>>seen >ACK's before in REGISTER processes, but I'm unfamiliar with 
>>the exact specifications of the SIP >protocol when it comes to 
>>REGISTERs.
>
>With non-INVITE transactions, you don't need to send an ACK. You 
>send a request,
>get a reply and that's it. Server doesn't know if client received the reply;
>that's not bad however, because client knows. The client keeps retransmitting
>until a reply arrives.
>
>INVITE is special-cased, since it takes long time until the remote 
>party answers.
>It thus makes no sense to solicit a final reply by retranmissions. That's the
>reason why client doesn't retransmit. To make sure that the final reply is
>in this case delivered, the retranmission job is carried out by server until
>ACK is received.
>
>Besides that, the "ACK" packet was broken -- it included fragments of several
>messages.
>
>-Jiri




>Date: Sun, 23 Feb 2003 22:44:10 +0100
>To: John Todd <jtodd at loligo.com>
>From: Jiri Kuthan <jiri at iptel.org>
>Subject: Re: Message from iptel.org SIP admin
>
>At 10:31 PM 2/23/2003, John Todd wrote:
>>Yes, the SIP support in Asterisk is... "rudimentary."
>>
>  >Can you forward a "good" register example so I can compare 
>side-by-side in the same format?
>
>There were some ok registrations coming from you too.
>
>
>>Is this causing problems?  If so, I can turn off my REGISTER service.
>
>Don't worry -- we just regularly observe our logs and report to users so
>that they can fix their problems.
>
>-Jiri
>
>a) proper REGISTER from asterisk
>
>#
>U 2003/02/23 01:05:44.719315 204.91.156.11:5060 -> 195.37.77.101:5060
>REGISTER sip:195.37.77.101 SIP/2.0.
>Via: SIP/2.0/UDP 204.91.156.11:5060;branch=457cbf4f.
>From: <sip:jtodd at 195.37.77.101>;tag=67eb0634.
>To: <sip:jtodd at 195.37.77.101>;tag=67eb0634.
>Contact: <sip:90764 at 204.91.156.11:5060;transport=udp>.
>Call-ID: 04c8ec626bcfae166db48a5367812a19 at 127.0.0.1.
>CSeq: 103 REGISTER.
>User-Agent: Asterisk PBX.
>Authorization: Digest username="jtodd", realm="iptel.org", 
>algorithm="MD5", uri="sip:jtodd at 195.37.77.101", 
>nonce="3e5811840000000074cb2641cd45675b9a68066afb405fd2", 
>response="e316f78b28fd221ccff47ce638225915".
>Expires: 120.
>Event: registration.
>
>#
>U 2003/02/23 01:05:44.721078 195.37.77.101:5060 -> 204.91.156.11:5060
>SIP/2.0 200 OK.
>Via: SIP/2.0/UDP 204.91.156.11:5060;branch=457cbf4f.
>From: <sip:jtodd at 195.37.77.101>;tag=67eb0634.
>To: <sip:jtodd at 195.37.77.101>;tag=67eb0634.
>Call-ID: 04c8ec626bcfae166db48a5367812a19 at 127.0.0.1.
>CSeq: 103 REGISTER.
>Contact: <sip:90764 at 204.91.156.11:5060;transport=udp>;q=0.00;expires=120.
>Server: Sip EXpress router (0.8.9 (i386/linux)).
>Content-Length: 0.
>Warning: 392 195.37.77.101:5060 "Noisy feedback tells: pid=7651 
>req_src_ip=204.91.156.11 in_uri=sip:195.37.77.101 
>out_uri=sip:195.37.77.101 via_cnt==1".
>.
>
>b) it is then followed by the following, broken message
>#
>U 2003/02/23 19:42:53.329964 204.91.156.10:5060 -> 195.37.77.101:5060
>ACK sip:jtodd at 195.37.77.101 SIP/2.0.
>Via: SIP/2.0/UDP 204.91.156.10:5060;branch=74ea2dbc.
>From: <sip:jtodd at 195.37.77.101>;tag=145e2c32.
>To: <sip:jtodd at 195.37.77.101>;tag=145e2c32.
>Contact: <sip:90764 at 204.91.156.10:5060;transport=udp>.
>Call-ID: 402d7f720d38fd477fdebfb37d3143f8 at 127.0.0.1.
>CSeq: 103 REGISTER.
>User-Agent: Asterisk PBX.
>Authorization: Digest username="jtodd", realm="iptel.org", 
>algorithm="MD5", uri="sip:jtodd at 195.37.77.101", 
>nonce="3e59175800000000d1858f7ad4424dfa92d3eace006eca13", 
>response="daa4e749f5b229247ed4b1c2b3d1ed5b".
>Expires: 120.
>Event: registration.
>.
>To: <sip:jtodd at 195.37.77.101>;tag=145e2c32.
>Contact: <sip:90764 at 204.91.156.10:5060;transport=udp>.
>Call-ID: 402d7f720d38fd477fdebfb37d3143f8 at 127.0.0.1.
>CSeq: 103 REGISTER.
>User-Agent: Asterisk PBX.
>Authorization: Digest username="jtodd", realm="iptel.org", 
>algorithm="MD5", uri="sip:jtodd at 195.37.77.101", 
>nonce="3e59175800000000d1858f7ad4424dfa92d3eace006eca13", 
>response="daa4e749f5b229247ed4b1c2b3d1ed5b".
>Expires: 120.
>Event: registration.
>.
>Call-ID: 402d7f720d38fd477fdebfb37d3143f8 at 127.0.0.1.
>CSeq: 103 REGISTER.
>User-Agent: Asterisk PBX.
>Authorization: Digest username="jtodd", realm="iptel.org", 
>algorithm="MD5", uri="sip:jtodd at 195.37.77.101", 
>nonce="3e59175800000000d1858f7ad4424dfa92d3eace006eca13", 
>response="daa4e749f5b229247ed4b1c2b3d1ed5b".
>Expires: 120.
>Event: registration.
>.
>CSeq: 103 ACK.
>User-Agent: Asterisk PBX.
>Content-Length: 0.
>.
>
>#
>
>c) proper Cisco Example
>
>#
>U 2003/02/23 01:05:14.556572 212.202.171.89:5060 -> 195.37.77.101:5060
>REGISTER sip:iptel.org SIP/2.0.
>Via: SIP/2.0/UDP 
>212.202.171.89:5060;branch=09495a5991b8b6e4a641eb08c20b0b6f.0.
>From: sip:jiri at iptel.org.
>To: sip:jiri at iptel.org.
>Call-ID: 00036bb9-0fd31922-54f72772-003f93fb at 192.168.2.33.
>Date: Sun, 23 Feb 2003 00:05:19 GMT.
>CSeq: 102 REGISTER.
>User-Agent: CSCO/4.
>Contact: <sip:jiri at 212.202.171.89>.
>Authorization: Digest 
>username="jiri",realm="iptel.org",uri="sip:iptel.org",response="7608172183cfc4191bd0c16a7cb90244",nonce="3e58116600000000269b864d1b5bcf21b31f0706e8cc2b5d",algorithm=MD5.
>Content-Length: 0.
>Expires: 600.
>.
>
>#
>U 2003/02/23 01:05:14.558092 195.37.77.101:5060 -> 212.202.171.89:5060
>SIP/2.0 200 OK.
>Via: SIP/2.0/UDP 
>212.202.171.89:5060;branch=09495a5991b8b6e4a641eb08c20b0b6f.0.
>From: sip:jiri at iptel.org.
>To: sip:jiri at iptel.org;tag=fb86ad2694115d75c77dce61523c9f07.bcd9.
>Call-ID: 00036bb9-0fd31922-54f72772-003f93fb at 192.168.2.33.
>CSeq: 102 REGISTER.
>Contact: <sip:jiri at 212.202.171.89>;q=0.00;expires=600.
>Server: Sip EXpress router (0.8.9 (i386/linux)).
>Content-Length: 0.
>Warning: 392 195.37.77.101:5060 "Noisy feedback tells: pid=7658 
>req_src_ip=212.202.171.89 in_uri=sip:iptel.org out_uri=sip:iptel.org 
>via_cnt==1".
>.




More information about the asterisk-users mailing list