[Home]

Summary:ASTERISK-02944: fmtp payload header
Reporter:sirs69 (sirs69)Labels:
Date Opened:2004-12-09 01:44:12.000-0600Date Closed:2011-06-07 14:10:17
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Nick Bachmann from astusers list thought I should submit this I'll make it quick: Basically a SIP provider we use mentioned to us that our fmtp range is 0-16 which should be 0-15 .. I have quote from:
http://www.cs.columbia.edu/~hgs/rtp/payload_audio.html
stating format should be set to 0-15, I could patch this however I wanted a few opinions on which is the 'standard'

****** ADDITIONAL INFORMATION ******

RTP Payloads for Telephone Signal Events
   RFC 2833
   Henning Schulzrinne, Scott Petrack.
   May 2000

   Implementation notes:

       * Implementations can support events 0 through 15 (DTMF) by simply ignoring the packets, but MUST declare all event numbers that are meaningful to it in the fmtp parameter, including 0 through 15.

sip.c (Thanks to Nick)

/* Indicate we support DTMF...  Not sure about 16, but MSN supports it
so dang it, we will too... */
                                 snprintf(costr, sizeof costr,
"a=fmtp:%d 0-16\r\n",


Comments:By: Mark Spencer (markster) 2004-12-09 09:58:01.000-0600

No, we really do want the 16, because that's flash hook. I went ahead and updated the comment and did the correct code to rtp.c as well.
By: Russell Bryant (russell) 2004-12-10 06:35:00.000-0600

made the same changes in 1.0 - will be in 1.0.4
By: Digium Subversion (svnbot) 2008-01-15 15:16:03.000-0600

Repository: asterisk
Revision: 4409

U   trunk/channels/chan_sip.c
U   trunk/rtp.c

------------------------------------------------------------------------
r4409 | markster | 2008-01-15 15:16:02 -0600 (Tue, 15 Jan 2008) | 2 lines

Update comment for fmtp 16, implement in RTP (bug ASTERISK-2944)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=4409
By: Digium Subversion (svnbot) 2008-01-15 15:16:07.000-0600

Repository: asterisk
Revision: 4414

U   branches/v1-0/channels/chan_sip.c

------------------------------------------------------------------------
r4414 | russell | 2008-01-15 15:16:07 -0600 (Tue, 15 Jan 2008) | 2 lines

update comment (bug ASTERISK-2944)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=4414
By: Digium Subversion (svnbot) 2008-01-15 15:16:08.000-0600

Repository: asterisk
Revision: 4415

U   branches/v1-0/rtp.c

------------------------------------------------------------------------
r4415 | russell | 2008-01-15 15:16:08 -0600 (Tue, 15 Jan 2008) | 2 lines

implement fmtp 16 in rtp (bug ASTERISK-2944)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=4415