[asterisk-bugs] [Asterisk 0011938]: Invalid interpretation of INVITE SIP frame
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Feb 6 13:12:10 CST 2008
The following issue has been set as RELATED TO issue 0011939.
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http://bugs.digium.com/view.php?id=11938
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Reported By: wilder
Assigned To: file
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Project: Asterisk
Issue ID: 11938
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.6.0-beta1
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: no change required
Fixed in Version:
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Date Submitted: 02-06-2008 10:05 CST
Last Modified: 02-06-2008 10:19 CST
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Summary: Invalid interpretation of INVITE SIP frame
Description:
Hello,
I have problem with trunk/1.6.0Beta2 Asterisk.
my sip.conf :
[XXXXXXXXX]
type=friend
username=XXXXXXXXX
fromuser=XXXXXXXXX
fromdomain=sip910.802.cz
secret=PASSWORD
host=212.71.146.172
context=802.cz-incoming
insecure=very
Everything works perfectly in Asterisk 1.4.17.
But when I use Asterisk TRUNK/1.6.0Beta2 incoming calls stop working.
I have analyzed SIP frames, I have found out, that Asterisk
TRUNK/1.6.0Beta2 has problems with handling INVITE frames...
Asterisk 1.4.17 that works correctly:
<--- SIP read from 212.71.146.172:5060 --->
INVITE sip:910800955 at 212.71.146.172 SIP/2.0
Call-ID: 4b8ed9fe-5196f9f8-17736357 at 212.71.146.172
Contact: <sip:421905400542 at 212.71.146.172>
CSeq: 102 INVITE
Expires: 1000
From: 421905400542
<sip:421905400542 at 212.71.146.172>;tag=a38c39a20c20a1ac19a17c16
To: <sip:910800955 at 212.71.146.172>
Via: SIP/2.0/UDP 212.71.146.172:5060;branch=z9hG4bK-47a9a54047a9d69f41
Max-Forwards: 70
Content-Type: application/sdp
Accept: application/sdp
User-Agent: PhoNetPbx
Content-Length: 284
v=0
o=PhoNetIP 78810735 78810735 IN IP4 212.71.146.184
s=SIP Call
c=IN IP4 212.71.146.184
t=0 0
a=sendrecv
m=audio 28939 RTP/AVP 18 2 8 101
a=fmtp:18 G729 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=ptime:20
<------------->
--- (13 headers 13 lines) ---
Sending to 212.71.146.172 : 5060 (NAT)
Using INVITE request as basis request -
4b8ed9fe-5196f9f8-17736357 at 212.71.146.172
Found peer '910800954'
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 212.71.146.184:28939
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x908
(alaw|g726|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 212.71.146.184:28939
Looking for 910800955 in 802.cz-incoming (domain 212.71.146.172)
list_route: hop: <sip:421905400542 at 212.71.146.172>
Asterisk TRUNK/1.6.0Beta2 that does not work :
<--- SIP read from UDP://212.71.146.172:5060 --->
INVITE sip:910800955 at 212.71.146.172 SIP/2.0
Call-ID: 4768d62d-5141f627-17750cbc at 212.71.146.172
Contact: <sip:421905400542 at 212.71.146.172>
CSeq: 102 INVITE
Expires: 1000
From: 421905400542
<sip:421905400542 at 212.71.146.172>;tag=a38c64a20c35a1ac27a17c21
To: <sip:910800955 at 212.71.146.172>
Via: SIP/2.0/UDP 212.71.146.172:5060;branch=z9hG4bK-47aaa6d747a9d7e5339
Max-Forwards: 70
Content-Type: application/sdp
Accept: application/sdp
User-Agent: PhoNetPbx
Content-Length: 284
v=0
o=PhoNetIP 78810820 78810820 IN IP4 212.71.146.178
s=SIP Call
c=IN IP4 212.71.146.178
t=0 0
a=sendrecv
m=audio 28762 RTP/AVP 18 2 8 101
a=fmtp:18 G729 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=ptime:20
<------------->
--- (13 headers 13 lines) ---
== Using SIP RTP CoS mark 5
Sending to 212.71.146.172 : 5060 (NAT)
Using INVITE request as basis request -
4768d62d-5141f627-17750cbc at 212.71.146.172
No user '421905400542' in SIP users list
Found peer '910800954' for '421905400542' from 212.71.146.172:5060
I attach whole SIP communication for Asterisk 1.4.17 that works correctly
and Asterisk 1.6.0Beta2 that doesn't work at all.
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Relationships ID Summary
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related to 0011939 The "contact" section of the ...
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Issue History
Date Modified Username Field Change
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02-06-08 13:12 putnopvut Relationship added related to 0011939
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