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Summary:ASTERISK-11389: Invalid interpretation of INVITE SIP frame
Reporter:Pavol Luptak (wilder)Labels:
Date Opened:2008-02-06 10:05:17.000-0600Date Closed:2011-06-07 14:02:49
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip-1.4.17.OK
( 1) sip-1.6.0-beta2.Error
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@212.71.146.172 SIP/2.0
Call-ID: 4b8ed9fe-5196f9f8-17736357@212.71.146.172
Contact: <sip:421905400542@212.71.146.172>
CSeq: 102 INVITE
Expires: 1000
From: 421905400542 <sip:421905400542@212.71.146.172>;tag=a38c39a20c20a1ac19a17c16
To: <sip:910800955@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@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@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@212.71.146.172 SIP/2.0
Call-ID: 4768d62d-5141f627-17750cbc@212.71.146.172
Contact: <sip:421905400542@212.71.146.172>
CSeq: 102 INVITE
Expires: 1000
From: 421905400542 <sip:421905400542@212.71.146.172>;tag=a38c64a20c35a1ac27a17c21
To: <sip:910800955@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@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.
Comments:By: Olle Johansson (oej) 2008-02-06 10:11:42.000-0600

You need to enable the debug channel too, and set core debugging to level 4 and capture so we see what's going on inside your asterisk. Thanks.
By: Joshua C. Colp (jcolp) 2008-02-06 10:19:01.000-0600

This is a configuration issue. "very" is no longer a valid option to insecure. Please replace it with:

insecure=invite,port