[asterisk-bugs] [Asterisk 0011750]: Asterisk ignores SDP when multiple Content-Type headers are present

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11750 
====================================================================== 
Reported By:                tasker
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11750
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.17 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             01-12-2008 09:56 CST
Last Modified:              01-12-2008 10:57 CST
====================================================================== 
Summary:                    Asterisk ignores SDP when multiple Content-Type
headers are present
Description: 
Asterisk ignores the SDP information when multiple Content-Type headers are
defined. Specifically, it seems to happen when SDP is not the first header
in a multiple Content-Type situation. The call is still handled and results
in one-way audio (inbound RTP only).
====================================================================== 

---------------------------------------------------------------------- 
 tasker - 01-12-08 10:57  
---------------------------------------------------------------------- 
<--- SIP read from 2.2.2.2:5060 --->
INVITE sip:3192908821 at 1.1.1.1;user=phone;cic=1234 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2
From: <sip:3097182212 at 2.2.2.2;user=phone>;tag=10000000-0-264876957
To: <sip:3192908821 at 1.1.1.1;user=phone>
CSeq: 1 INVITE
MIME-Version: 1.0
Content-Length: 510
Contact: <sip:3097182212 at 2.2.2.2>
Call-ID: 12767973 at 2.2.2.2
Remote-Party-ID:
<sip:3097182212 at 2.2.2.2;user=phone>;party=calling;id-type=subscriber;privacy=off;screen=no
Max-Forwards: 70
Content-Type: multipart/mixed; boundary=Telica-boundary

--Telica-boundary
Content-Type: application/sdp
Content-Disposition: session; handling=required

v=0
o=- 5187263383 5187263383 IN IP4 2.2.2.2
s=-
c=IN IP4 2.2.2.2
t=0 0
m=audio 14442 RTP/AVP 0 101
a=sendrecv
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--Telica-boundary
Content-Type: application/ISUP; version=ansi00
Content-Disposition: signal; handling=required


¢#! RG#"!C
<------------->
--- (12 headers 22 lines) ---
Sending to 2.2.2.2 : 5060 (no NAT)
Using INVITE request as basis request - 12767973 at 2.2.2.2
Found no matching peer or user for '2.2.2.2:5060'
Looking for 3192908821 in trisun (domain 1.1.1.1)
list_route: hop: <sip:3097182212 at 2.2.2.2>

<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;received=2.2.2.2
From: <sip:3097182212 at 2.2.2.2;user=phone>;tag=10000000-0-264876957
To: <sip:3192908821 at 1.1.1.1;user=phone>
Call-ID: 12767973 at 2.2.2.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:3192908821 at 1.1.1.1>
Content-Length: 0


<------------>
Audio is at 1.1.1.1 port 18926
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
courier*CLI>
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;received=2.2.2.2
From: <sip:3097182212 at 2.2.2.2;user=phone>;tag=10000000-0-264876957
To: <sip:3192908821 at 1.1.1.1;user=phone>;tag=as72e07e50
Call-ID: 12767973 at 2.2.2.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:3192908821 at 1.1.1.1>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 22521 22521 IN IP4 1.1.1.1
s=session
c=IN IP4 1.1.1.1
t=0 0
m=audio 18926 RTP/AVP 18 0 3
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
courier*CLI>
<--- SIP read from 2.2.2.2:5060 --->
ACK sip:3192908821 at 1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2
Call-ID: 12767973 at 2.2.2.2
From: <sip:3097182212 at 2.2.2.2;user=phone>;tag=10000000-0-264876957
To: <sip:3192908821 at 1.1.1.1;user=phone>;tag=as72e07e50
CSeq: 1 ACK
Contact: <sip:3097182212 at 2.2.2.2>
Content-Length: 0
Max-Forwards: 70


<------------->
--- (9 headers 0 lines) ---
courier*CLI> 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-12-08 10:57  tasker         Note Added: 0076808                          
======================================================================