[asterisk-bugs] [Asterisk 0015586]: Failure to negotiate T.38

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jul 30 09:49:35 CDT 2009


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15586 
====================================================================== 
Reported By:                globalnetinc
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15586
Category:                   Codecs/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-07-26 15:57 CDT
Last Modified:              2009-07-30 09:49 CDT
====================================================================== 
Summary:                    Failure to negotiate T.38
Description: 
To implement T.38 on most ATAs their is a reinvite required.  In the
process of gatewaying the T.38 negotiations the Asterisk server is not
doing this correctly.  On versions past 1.6.0.10 it does not even send the
same ports on the RTP streams to both parties.  

Every version past 1.6.0.10 fails
1.6.0.11
1.6.1.0
1.6.1.1
1.6.2.0-rc

This also includes the new T.38 stack that is is being introduced. in the
SVN tree of 1.6.1.1 and 1.6.2.0
====================================================================== 

---------------------------------------------------------------------- 
 (0108394) globalnetinc (reporter) - 2009-07-30 09:49
 https://issues.asterisk.org/view.php?id=15586#c108394 
---------------------------------------------------------------------- 
Here is an example in 1.6.2-beta3

INVITE sip:4065877430 at 216.166.170.179:5060 SIP/2.0M
Via: SIP/2.0/UDP 216.166.168.6:5060;branch=z9hG4bK0e1a10f6;rportM
Max-Forwards: 70M
From: <sip:5864348 at 216.166.168.6>;tag=as5701e186M
To: The Johnson Company
<sip:4065877430 at 216.166.168.6>;tag=dfd89b08f93f9771o1M
Contact: <sip:5864348 at 216.166.168.6>M
Call-ID: d1eec814-13e26dd5 at 216.166.170.179M
CSeq: 103 INVITEM
User-Agent: Asterisk PBXM
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOM
Supported: replaces, timerM
X-asterisk-Info: SIP re-invite (External RTP bridge)M
Content-Type: application/sdpM
Content-Length: 264M
M
v=0M
o=root 1412831342 1412831346 IN IP4 66.62.196.38M
s=Asterisk PBXM
c=IN IP4 66.62.196.38M
t=0 0M
m=image 14648 udptl t38M
a=T38Faxversion:0M
a=T38MaxBitRate:14400M
a=T38FaxRateManagement:transferredTCFM
a=T38FaxMaxDatagram:165M
a=T38FaxUdpEC:t38UDPRedundancyM

---
[Jul 30 00:12:07] VERBOSE[24901] logger.c:
<--- SIP read from UDP://216.166.170.179:5060 --->
SIP/2.0 200 OKM
To: The Johnson Company
<sip:4065877430 at 216.166.168.6>;tag=dfd89b08f93f9771o1M
From: <sip:5864348 at 216.166.168.6>;tag=as5701e186M
Call-ID: d1eec814-13e26dd5 at 216.166.170.179M
CSeq: 103 INVITEM
Via: SIP/2.0/UDP 216.166.168.6:5060;branch=z9hG4bK0e1a10f6M
Contact: The Johnson Company <sip:4065877430 at 216.166.170.179:5060>M
Server: Linksys/SPA2102-5.2.5M
Content-Length: 275M
Content-Type: application/sdpM
M
v=0M
o=- 22383364 22383364 IN IP4 216.166.170.179M
s=-M
c=IN IP4 216.166.170.179M
t=0 0M
m=image 16394 udptl t38M
a=T38FaxVersion:0M
a=T38MaxBitRate:14400M
a=T38FaxRateManagement:transferredTCFM
a=T38FaxMaxBuffer:200M
a=T38FaxMaxDatagram:200M
a=T38FaxUdpEC:t38UDPRedundancyM

<------------->
[Jul 30 00:12:07] VERBOSE[24901] logger.c: --- (10 headers 12 lines) ---
[Jul 30 00:12:07] VERBOSE[24901] logger.c: Got T.38 offer in SDP in dialog
d1eec814-13e26dd5 at 216.166.170.179
[Jul 30 00:12:07] VERBOSE[24901] logger.c: Got T.38 Re-invite without
audio. Keeping RTP active during T.38 session. Callid
d1eec814-13e26dd5 at 216.166.170.179
[Jul 30 00:12:07] VERBOSE[24901] logger.c: Capabilities: us - 0x114
(ulaw|g729|g726), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0
(nothing), combined - 0x0 (nothing)
[Jul 30 00:12:07] VERBOSE[24901] logger.c: Non-codec capabilities (dtmf):
us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jul 30 00:12:07] VERBOSE[24901] logger.c: set_destination: Parsing
<sip:4065877430 at 216.166.170.179:5060> for address/port to send to
[Jul 30 00:12:07] VERBOSE[24901] logger.c: set_destination: set
destination to 216.166.170.179, port 5060
[Jul 30 00:12:07] VERBOSE[24901] logger.c: Transmitting (no NAT) to
216.166.170.179:5060:

Then you see this error:
WARNING[16385] udptl.c: UDPTL asked to send 41 bytes of IFP when far end
only prepared to accept 31 bytes; data loss may occur. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-30 09:49 globalnetinc   Note Added: 0108394                          
======================================================================




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