[asterisk-bugs] [Asterisk 0013008]: [patch] chan_sip ignores rport and does not reply to source IP:port
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Jul 8 06:55:47 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=13008
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Reported By: klaus3000
Assigned To: bbryant
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Project: Asterisk
Issue ID: 13008
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
Asterisk Version: 1.4.21
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 07-07-2008 08:44 CDT
Last Modified: 07-08-2008 06:55 CDT
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Summary: [patch] chan_sip ignores rport and does not reply to
source IP:port
Description:
Hi!
I have defined a peer in sip.conf:
[klaus]
type=peer
username=klaus
secret=password
host=dynamic
context=fromklaus
Then I register with eyebeam. Eyebeam sets the "rport" parameter in via
header which should activate symmetric signaling. Further, even if rport is
missing, the RFC 3261 mandates that replies should be sent to the source IP
of the request, not the IP in the Via header (the latter one should only be
used if the first one fails (e.g. ICMP error)).
Bug: Asterisk sends the request back to the IP in the Via header. I know I
can change behavior by setting nat=yes, but even without this setting
Asterisk should be standard conform.
I used Asterisk 1.4.21.1
======================================================================
----------------------------------------------------------------------
klaus3000 - 07-08-08 06:55
----------------------------------------------------------------------
Trace + config:
<--- SIP read from 83.136.33.3:10622 --->
REGISTER sip:81.16.157.161 SIP/2.0
Via: SIP/2.0/UDP
10.10.0.51:10622;branch=z9hG4bK-d8754z-a6180f2e6c50f958-1---d8754z-;rport
Max-Forwards: 70
Contact:
<sip:klaus at 83.136.33.3:10622;rinstance=a6b3da612a4bcbb0;transport=udp>
To: <sip:klaus at 81.16.157.161>
From: <sip:klaus at 81.16.157.161>;tag=2b240f47
Call-ID: ZTJlN2Y5OTljMzQwZmRmN2M4OWMyY2RjYWE1ZTc4NjM.
CSeq: 1 REGISTER
Expires: 72000
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 83.136.33.3 : 10622 (NAT)
<--- Transmitting (no NAT) to 10.10.0.51:10622 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.10.0.51:10622;branch=z9hG4bK-d8754z-a6180f2e6c50f958-1---d8754z-;received=83.136.33.3;rport=10622
From: <sip:klaus at 81.16.157.161>;tag=2b240f47
To: <sip:klaus at 81.16.157.161>
Call-ID: ZTJlN2Y5OTljMzQwZmRmN2M4OWMyY2RjYWE1ZTc4NjM.
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:klaus at 81.16.157.161>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.10.0.51:10622 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.10.0.51:10622;branch=z9hG4bK-d8754z-a6180f2e6c50f958-1---d8754z-;received=83.136.33.3;rport=10622
From: <sip:klaus at 81.16.157.161>;tag=2b240f47
To: <sip:klaus at 81.16.157.161>;tag=as0de19ba4
Call-ID: ZTJlN2Y5OTljMzQwZmRmN2M4OWMyY2RjYWE1ZTc4NjM.
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="2fb40a83"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'ZTJlN2Y5OTljMzQwZmRmN2M4OWMyY2RjYWE1ZTc4NjM.' in 32000 ms (Method:
REGISTER)
sip show settings
Global Settings:
----------------
SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
Always auth rejects: No
Call limit peers only: No
Direct RTP setup: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS SIP: none
IP ToS RTP audio: none
IP ToS RTP video: none
T38 fax pt UDPTL: No
RFC2833 Compensation: No
SIP realtime: Disabled
Global Signalling Settings:
---------------------------
Codecs: 0x8000e (gsm|ulaw|alaw|h263)
Codec Order: none
T1 minimum: 100
Relax DTMF: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Default Settings:
-----------------
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
I do not have any nat=... directive in sip.conf (thus default value should
be used).
btw: What is nat=never exactly? Does it only ignore rport parameter (RFC
3581) or does ot also ignores received parameter (RFC 3261)?
Issue History
Date Modified Username Field Change
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07-08-08 06:55 klaus3000 Note Added: 0089882
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