Wireshark-dev: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark
From: Dietfrid Mali <karx11erx@xxxxxxxxxxx>
Date: Wed, 26 Jan 2011 09:23:58 +0100
Yeah, I wasn't expressing this quite right. What I basically wanted to say is that Wireshark
doesn't assign a payload type using the SDP but just forwards the payload type
given in the RTP packets (102, which Wireshark wraps into a define that basically says
"undefined packet type").

> Date: Wed, 26 Jan 2011 00:09:50 +0100
> From: jaap.keuter@xxxxxxxxx
> To: wireshark-dev@xxxxxxxxxxxxx
> Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark
>
> On 01/25/2011 05:48 PM, Dietfrid Mali wrote:
> > The problem with e.g. G.726 is that Wireshark gives those packets RTP
> > type 102 which afaik is an error code ("unknown encoding").
>
> No, that's your RTP endpoint configured to label these as such. RFC 3550 says:
> "A profile MAY specify a default static mapping of payload type codes to payload
> formats. Additional payload type codes MAY be defined dynamically through
> non-RTP means (see Section 3)."
> RFC 1890/RFC 3551 defines the "RTP Profile for Audio and Video Conferences with
> Minimal Control", which lists several static payload types. The old RFC lists
> G.721 (aka G.726-32), while the new one dropped that one and added references to
> G.726 at various bit rate, with a dynamic payload type.
> RFC 3550 says in Section 3: "Non-RTP means: Protocols and mechanisms that may be
> needed in addition to RTP to provide a usable service. In particular, ..., and
> define dynamic mappings between RTP payload type values and the payload formats
> they represent for formats that do not have a predefined payload type value."
> with reference to Session Description Protocol (SDP)
>
> So, payload type 102 is a dynamic payload type which has to be given meaning
> (through SDP for instance) within the session. In your case Wireshark didn't
> pick that up from the trace, hence cannot give you the proper interpretation of
> that payload type within that session.
>
> > I would need to know where and how Wireshark maps dynamic payload types
> > (negotiated via SDP) to internal static ones. Above that RFC3551 notes that
> > static G.726 payload types are obsolete, and afaik there aren't even
> > (obsolete) static payload types for all G.726 variants, so Wireshark
> > would need to
> > take care of that by using some (more or less arbitrary) internal static
> > type numbers.
>
> Yep, that is done by the SDP dissector. It tries to interpret the SDP offer
> (should be the answer, but that a whole other story) and create conversations
> (see doc/README.developer, section 2.2) for the RTP dissector, feeding it
> dynamic payload type information it has learned from the media attributes.
>
> The RTP dissector does then the heavy lifting on the RTP packets, based on the
> information feed in by the SDP dissector.
>
> > I will do my best to provide a patch once I have fully integrated all
> > codecs (currently only G.726-32 has been implemented as proof of
> > concept, but since
> > this is working adding more is no big deal).
>
> Just one to get started is fine. Does it integrate into codecs/ directory
> besides G711a and G711u (and G729 and G723, if you have them)?
>
> > Getting G.726 to work was a bit of a pain btw because of the weird frame
> > sync calculation in rtp_player.c::play_channels() as this function seems to
> > assume 1:1 relationships of decoder input and output stream sizes and
> > thus simply halves the decoder output batch sizes to determine whether
> > frames
> > are properly sync'd. This doesn't work for compressed audio. To
> > compensate, my decodeG726_32() function doubles the number of bytes returned
> > (as it has a 1:2 relationship of input and output buffer sizes). Before
> > it did that, lots of silence frames were inserted and half of the audio
> > data was
> > dropped by the player.
>

The G.726 decoder function returns twice the number of bytes fed to it as
output buffer size. Just returning that number leads to incorrect audio
playback. I had to double that number once more to make it work. This has something
to do with how rtp_player() determines whether it has received enough packets
for the given time frame and that it inserts silence frames when it thinks it hasn't.
There is no (inline) documentation of what rtp_player() expects. Maybe this is
described in some Wireshark programming or API documentation?

> I'm no sure if I understand you correctly. Working with these decode functions
> there is an input buffer with its length as input, and two output parameters,
> being the output buffer and it a pointer to store its size. This size of the
> output buffer has to be set, by the decoder, to the number of samples in output
> buffer. That should be enough, see for instance rtp_player.c:decode_rtp_packet()
> the handling of G.279 and G.723.
> Be aware that you have to store 16 bit linear samples in the output buffer,
> maybe that's your factor 2?
>
> Thanks,
> Jaap
>
>
> >
> > Dietfrid
> >
> > > From: jaap.keuter@xxxxxxxxx
> > > Date: Tue, 25 Jan 2011 16:54:25 +0100
> > > To: wireshark-dev@xxxxxxxxxxxxx
> > > Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark
> > >
> > > Hi,
> > >
> > > That would be interesting. Can you put the code in a patch on bugzilla?
> > >
> > > Can't work on it right now, but would be nice to have.
> > >
> > > btw: their are already static RTP types assigned for both codecs. The
> > dynamic types should come in through protocols like SDP, or a dissector
> > preference.
> > >
> > > Thanks,
> > > Jaap
> > >
> > > Send from my iPhone
> > >
> > > On 25 jan. 2011, at 16:07, Dietfrid Mali <karx11erx@xxxxxxxxxxx> wrote:
> > >
> > > > Hi,
> > > >
> > > > using spandsp, I have added G.722 and G.726 decoders to Wireshark.
> > > >
> > > > Currently this is a bit of a hack job, particularly regarding
> > inclusion of the spandsp lib, and I could need a bit help to properly
> > integrate it into Wireshark's automake hell (configure.in).
> > > >
> > > > There also isn't a proper Wireshark signature for that RTP type (I
> > am simply reacting to RTP type 102, which actually is an error code), so
> > some help getting this straight and introducing proper codec types would
> > be appreciated, too.
> > > >
> > > >
> > > >
>
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