Summary:ASTERISK-18859: Unable to place calls from some SIP phones ("Multiple audio streams are not supported")
Reporter:Birger "WIMPy" Harzenetter (wimpy)Labels:
Date Opened:2011-11-14 07:38:08.000-0600Date Closed:2011-11-16 15:21:20.000-0600
Versions:SVN Frequency of
is related toASTERISK-18749 (Only) First attempt to put a call on hold fails when using SRTP
Environment:Attachments:( 0) sipdebug
Description:Calls are rejected immediately with
WARNING[17791]: chan_sip.c:9023 process_sdp: Multiple audio streams are not supported
SIP/2.0 488 Not acceptable here
Comments:By: Birger "WIMPy" Harzenetter (wimpy) 2011-11-14 07:38:45.864-0600

sip debug of failed call attempt

By: David Woolley (davidw) 2011-11-14 08:31:58.378-0600

Not a bug.  Asterisk doesn't claim to support this.

The remote device seems to be making conflicting bids for the same port number, with and without encryption.  It could never hope to have both of them succeed.

The correct way of doing this would seem to be RFC5939, but I don't think Asterisk supports that either.

By: Birger "WIMPy" Harzenetter (wimpy) 2011-11-14 08:46:07.878-0600

Thanks for the fast reply.
I have no idea, it the implementation to offer either an encrypted OR an unencrypted audio stream is correct. It used to work so far.
But changing 'user_savp' from 'optional' to 'mandatory' on the phone fixed the issue.
Thanks again

By: David Vossel (dvossel) 2011-11-14 10:28:39.210-0600

If this used to work, it should continue to work.

By: David Woolley (davidw) 2011-11-14 11:05:46.762-0600

Earlier versions didn't have a specific check for multiple audio streams, so would simply have mis-parsed the SDP.

On a quick skim, it looks to me as though would have used the last found audio stream, although it might have mixed in attributes from earlier ones as well. does actually check for too many streams, but the test only works if you have four or more streams, as it is assuming that the first three will be: first audio; video; and text streams.

By: David Vossel (dvossel) 2011-11-14 11:26:42.702-0600

Birger, that SDP is odd.  Is this snom's way of doing some sort of best effort SRTP?  The way it is being done is completely insecure.  Is it possible to get the snom to do SRTP without sending out the offer for the unencrypted stream?

By: Birger "WIMPy" Harzenetter (wimpy) 2011-11-14 11:54:45.060-0600

Yes. That's a best effort setting.
I used that when I started testing SRTP so I only had to change Asterisks configuration.
And yes again it's possible to use only one offer. That's how I fixed it as per my first comment.
see also http://wiki.snom.com/wiki/index.php/Settings/user_savp

So it seems to come down to a simple Don't use that setting any more.

By: Birger "WIMPy" Harzenetter (wimpy) 2011-11-14 12:54:33.866-0600

setting RTP/SAVP to mandatory also fixes ASTERISK-18749

By: Matt Jordan (mjordan) 2011-11-16 15:21:11.587-0600

I'm going to go ahead and close this issue, as a workaround is available to resolve the SRTP settings on the snom phone.

At this point in time, Asterisk does not support the concept of multiple streams of the same type.  Doing so would be a useful enhancement, but would be a new feature request for Asterisk.  If you'd like to discuss having this feature added, the proper forum would be a discussion on the Asterisk mailing lists [1] and IRC channels.

[1] http://www.asterisk.org/support/mailing-lists