Is possibile to have a on_rx_rtp callback with PJSUA?

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



On 3/19/08, Roland Klabunde <roland.klabunde at freenet.de> wrote:
> > So you want to do some signal analysis. The good news is that the
>  > media has been established, so we can just peek at the PCM signal
>  > rather than hacking the incoming RTP packets, and this is very easy
>  > with PJSUA-LIB!
>
> I'm not up to date here. What in case of G.72x? Does he have access to plain
>  PCM data? If so, the signal analysis isn't really that impossible, I claimed
>  before :)

Since we've got 183 response with SDP, that means codec has been
negotiated, so we don't need to care about RTP and just work at PCM
level.

>  But what in case the user is busy? Or he gets some sort of voice annotation?
>  ("Then number you have dialed...") Could become more problematic than,
>  though

Well I didn't mean signal analysis is easy. ;-)
What I mean to say was that if you want to peek at the PCM samples,
that should be easy with PJSUA-LIB. For the signal analysis itself,
I'll leave that to Davide. :)

cheers,
 -benny


>  Regards
>
>
>
>
>  ----- Original Message -----
>  From: "Benny Prijono" <bennylp@xxxxxxxxx>
>  To: "pjsip list" <pjsip at lists.pjsip.org>
>  Sent: Wednesday, March 19, 2008 6:45 PM
>  Subject: Re: Is possibile to have a on_rx_rtp callback with PJSUA?
>
>
>
> > On 3/19/08, Davide Marrone <unidavide at email.it> wrote:
>  >> Roland Klabunde wrote:
>  >>  >> The the problem is: how can I understand from the statistics when the
>  >>  >> remote phone is ringing ?
>  >>  > If there is no explicit RINGING SIP message, you would have to analyze
>  >> the
>  >>  > received RTP in order to find the ringtone pattern... Nearly
>  >> impossible, I
>  >>  > guess.
>  >>
>  >> Is exactly what I want to do, I want to analyze the RTP traffic and
>  >>  recognize the ringtone pattern. Firt of all I need to get the RTP
>  >>  traffic, have you any suggestion to get it?
>  >
>  > Ah right, I misunderstood what you're trying to do (and it seems that
>  > we've been discussing the wrong topic!).
>  >
>  > So you want to do some signal analysis. The good news is that the
>  > media has been established, so we can just peek at the PCM signal
>  > rather than hacking the incoming RTP packets, and this is very easy
>  > with PJSUA-LIB!
>  >
>  > What you need to do is something like this:
>  > - implement your signal analysis as a sink media port
>  > (http://trac.pjsip.org/repos/wiki/FAQ#audio-man)
>  > - register this port to pjsua-lib's conference bridge
>  > (pjsua_conf_add_port())
>  > - once call's media is establish, connect the call's media slot to
>  > your signal analysis port (with pjsua_conf_connect()) and begin your
>  > signal analysis!
>  >
>  > Cheers,
>  > -benny
>  >
>
> > _______________________________________________
>  > Visit our blog: http://blog.pjsip.org
>  >
>  > pjsip mailing list
>  > pjsip at lists.pjsip.org
>  > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>  >
>
>
>  _______________________________________________
>  Visit our blog: http://blog.pjsip.org
>
>  pjsip mailing list
>  pjsip at lists.pjsip.org
>  http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>



[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux