[asterisk-bugs] [Asterisk 0019281]: [patch] Invite with session description that supports both SRTP(SAVP) and RTP(AVP) fails if asterisk is not configured to use SR

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=19281 
====================================================================== 
Reported By:                jacco
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   19281
Category:                   Channels/chan_sip/SRTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.4 
JIRA:                       SWP-3458 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-05-12 02:17 CDT
Last Modified:              2011-05-12 09:48 CDT
====================================================================== 
Summary:                    [patch] Invite with session description that
supports both SRTP(SAVP) and RTP(AVP) fails if asterisk is not configured to use
SR
Description: 
I was doing an interoperability with a HiPath 3000 V8 M5T SIP
Stack/4.0.26.26 and it failed.

It has to do with an invite with session desription that supports both
SRTP(SAVP) and RTP(AVP); when asterisk is not configured for SRTP it will
not start an RTP session.
Asterisk CLI shows : 
chan_sip.c: Can't provide secure audio requested in SDP offer
and does not continue with setting up rtp

I guess it should provide an insecure audio request en continue with RTP

tested in asterisk 1.8.2.4 and asterisk 1.8.4



====================================================================== 

---------------------------------------------------------------------- 
 (0134839) jacco (reporter) - 2011-05-12 09:48
 https://issues.asterisk.org/view.php?id=19281#c134839 
---------------------------------------------------------------------- 
I've got a response back from siemens:
L.S.,

In my opinion, the SDP offered by the Hipath is correct according the
RFC3264.

The offer will contain zero or more media streams (each media stream
   is described by an "m=" line and its associated attributes).  

A port number of zero in the offer indicates that the
   stream is offered but MUST NOT be used.  This has no useful semantics
   in an initial offer, but is allowed for reasons of completeness,
   since the answer can contain a zero port indicating a rejected stream

In all cases, the formats in the "m=" line MUST be listed in order of
   preference, with the first format listed being preferred.  In this
   case, preferred means that the recipient of the offer SHOULD use the
   format with the highest preference that is acceptable to it.

If multiple media streams of different types are present, it means
   that the offerer wishes to use those streams at the same time.  A
   typical case is an audio and a video stream as part of a
   videoconference.


v=0
o=MxSIP 0 1849763221 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 29100 RTP/AVP 8 0 18 4 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
a=sendrecv
m=audio 0 RTP/SAVP 8 0 18 4 98 99
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 telephone-event/8000
a=rtpmap:99 red/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=fmtp:99 98
a=sendrecv

The offered SDP suggests 2 audio streams, the second M-line has a port
ZERO set and therefore MUST NOT be used.

The response also MUST contain both M-lines. 
The second M-line can be rejected in the response by setting the port
ZERO.
Therefore the Server: Asterisk PBX 1.8.2.4 MUST respond to the SDP with an
answer if one of the 2 m-lines is supported by the Server: Asterisk PBX
1.8.2.4. 
If none of the m-lines is supported. The 488 answer is correct. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-12 09:48 jacco          Note Added: 0134839                          
======================================================================