Hi Davide,

I don't think it works like you described.
When your client receive a call, the callback "on_incoming_call" is called.
You must reply with msg 180 :
pjsua_call_answer(call_id, 180, NULL, NULL);
If you reply immediately by msg 200, the call is established.

To reply you must call the following function to send msg 200 :
pjsua_call_answer(current_call, 200, NULL, NULL);

Samuel

Davide Marrone a écrit :

Hi Samuel,
maybe my question wasn't exactly clear. I don't need to receive or hang
up an incoming call, I need to make an outgoing call from my sip client
to exeternal phones. The problem is that I don't know exatly when the
external phone is ringing because via SIP aren't sent packets about that.

Davide

Samuel Vinson wrote:

On 3/15/08, Davide Marrone unidavide@email.it wrote:

Is this because the 18x response is lost? If so, reliable provisional
response should solve it I think. You can require remote to send
response reliably with --use-100rel option in pjsua.

If you think that listening for incoming RTP is the only way to go,
then for now the only way is to hack the UDP media transport to call
your callback whenever there's an incoming RTP packet. Or something
like that, I haven't thought of this very thoroughly.

In the future I'm planning to add another callback in pjsua-lib to
allow application to create its own media transport (or to report
media transport creation). This among other thing will enable
application such as yours to do some work with the transport.

cheers,
-benny

Hi Benny,
first of all thanks for your reply

Benny Prijono wrote:

no, the problem isn't that, the service (voip.eutelia.it and others
italian services) doesn't send this packet, I don't know why, but it
doesn't arrive, if I use --use-100rel when I try to make a call I get

20:18:40.527  test.c Entered in on_call_state
20:18:40.527  test.c Call 0 is DISCONNECTED [reason=420 (Bad Extension)]

while without it the call is made correctly, anyway I don't think that
100rel is the problem

Yes, I'm really sure that the only thing to do is to intercept the
incoming RTP traffic. I have discoverd it using xten-xlite, I have made
a lot of tries with tcpdump & ngrep, when I make a call to an external
phone I have saw that:

• the SIP packet 180 ringing doesn't arrive, never
• I receive a SIP packet: 183 Session Progress
• xten-xlite doesn't play the local ringing tone until the change of RTP
  traffic

I really made many tries and I'm sure that the only SIP packet that
arrive when the call in on going is 183 Session Progress and "viewing"
with ngrep the RTP packets is really clear that, at a moment when the
traffic change, the client start to play the local ringing tone that
simulate the remote ringing. This moment is exactly when the remote
phone is really ringing. So I'm sure at 100% that I have to get the RTP
traffic.
I don't already know what changes of the traffic I have to intercept but
first of all I have to get it.

great, but for now there is a solution to my problem?
I have tried to do that:

// ...
// initialization of pjsua and registration to the server
// ...

// make the call
status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, &call_id);
if (status != PJ_SUCCESS) error_exit("Error making call", status);
		
// the call start correctly, i receive the SIP 183 packet
// and my remote phone rings

pjsua_call *call;
pjmedia_sock_info skinfo;

call = &pjsua_var.calls[call_id];		

// get transport info
	pjmedia_transport_get_info(call->med_tp, &skinfo);
if (status != PJ_SUCCESS) error_exit("Error getting info", status);

pj_sockaddr_in rem_addr;
pj_sockaddr_in rem_rtcp;
rem_addr= (pj_sockaddr_in)skinfo.rtp_addr_name;
rem_rtcp= (pj_sockaddr_in)skinfo.rtcp_addr_name;

// is correct this detach ?
pjmedia_transport_detach(call->med_tp, call->user_data);
if (status != PJ_SUCCESS) error_exit("Error detaching transport", status);

// is correct this ?
pjmedia_transport_attach(call->med_tp, NULL, &rem_addr, &rem_rtcp, 

sizeof(pj_sockaddr_in), &on_rx_rtp, NULL);
if (status != PJ_SUCCESS) error_exit("Error attaching transport", status);

but it doesn't work, I'm using a modded version of pjsua, but when I run
it I got

20:28:58.792 os_core_unix.c pjlib 0.8.0 for POSIX initialized
20:28:58.792 sip_endpoint.c Creating endpoint instance...
20:28:58.793          pjlib select() I/O Queue created (0x814956c)
20:28:58.793 sip_endpoint.c Module "mod-msg-print" registered
20:28:58.793 sip_transport. Transport manager created.
20:28:58.809  pjsua_core.c pjsua version 0.8.0 for i686-pc-linux-gnu
initialized
20:28:58.898    pjsua_acc.c Registration sent
20:28:59.138    pjsua_acc.c sip:024XXXXX@voip.eutelia.it:
unregistration success
20:28:59.138        test.c Entered in on_reg_state
20:28:59.138        test.c Successfully logged in entering main loop
20:28:59.208        test.c Entered in on_call_state
20:28:59.208        test.c Call 0 state changed to CALLING
20:28:59.208        test.c Entered in on_call_tsx_state
20:28:59.438        test.c Entered in on_call_state
20:28:59.438        test.c Call 0 state changed to CALLING
20:28:59.438        test.c Entered in on_call_tsx_state
20:28:59.438        test.c Entered in on_call_tsx_state
20:28:59.592        test.c Entered in on_call_tsx_state
20:29:01.359        test.c Entered in on_call_state
20:29:01.359        test.c Wait and cancel call numeber: 0
20:29:01.359        test.c Call 0 state changed to EARLY (183 Session
Progress)
test: ../src/pjmedia/transport_udp.c:561: transport_attach: Assertion
`!udp->attached' failed.
Aborted

what's wrong with my code? I have put the pjmedia_transport_detach in
the code but I got that error.

thanks
Davide

p.s. anyway Benny my best congratulations for all you work, this
framework is really incredible and useful !

On 3/17/08, Davide Marrone unidavide@email.it wrote:

Yeah, the 420 response means remote doesn't support the extension.

Sorry now I'm a bit lost. If you said you've got 183, then why can't
you just "start the call" with the arrival of this 183 response
(rather than waiting for the 180 response). It is quite common to
receive 183 instead of 180 if you are calling a PSTN gateway, and if
you listen to the incoming audio most likely you will hear ringing
tone with the 183 response.

Since you've had 183 (and I assume there is SDP in the 183), your
media has been established, and your on_call_media_state() callback
should have been called.

So now you can just poll the media statistic of the call to see if
you've received RTP packets. You can use "dq" command in pjsua to show
the call's media statistic (and trace the implementation of this
command to see how to retrieve the media statistic programmatically).

I'm not sure why your code below doesn't work though.

cheers,
-benny

Benny Prijono wrote:

Yes, I have to call mobile phones.

Yes, you right, in the 183 there is SDP, the media is established and
the on_call_media_state is called.

I have tried to do that, I saw that the "dq" calls the pjsua_call_dump
and I saw in the implementation that it calls dump_media_session to
retrive RTP statistic.
The the problem is: how can I understand from the statistics when the
remote phone is ringing ?
I have to know exactly when the remote phone start to ring and I have to
immediatly hang up the call.
The program that I'm writing must do only a single ring on the remote
phone, this is a kind of "cover signal" that wakeup the mobile phone.

The program need to be perfectly synchronized because I have to minimize
the ringing time so the user cannot answer to the call. If the user
answer to the call I have a to pay for it, so the ringing time must be
really short to avoid charge for me.

cheers
Davide

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.

----- Original Message -----
From: "Davide Marrone" unidavide@email.it
To: "pjsip list" pjsip@lists.pjsip.org
Sent: Wednesday, March 19, 2008 3:45 PM
Subject: Re: [pjsip] Is possibile to have a on_rx_rtp callback with PJSUA?

Roland Klabunde wrote:

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?

cheers
Davide

On 3/19/08, Roland Klabunde roland.klabunde@freenet.de wrote:

David,

Since originally you just want to know whether incoming RTP packets
have been received, you can just check that in the call media
statistic (it's there somewhere if you look at it carefully ;-) ).

But Roland is right, just because you have RTP doesn't mean remote is
ringing. It could well be that remote is sending "Sorry the number is
wrong" prompt.

Cheers,
-benny

Benny Prijono wrote:

Hi Benny,
in my first message I wrote:

maybe wasn't exactly clear, sorry for my english, I know that is not so
good.
Now I think that all is clear, I saw the media statistics and I saw the
number of incoming RTP packets but, you have correctly said that RTP
doesn't mean remote is ringing, so I need to get the RTP traffic and
analyze it.

Can you give me some hint to get the RTP traffic?

thanks again
Davide

On 3/19/08, Davide Marrone unidavide@email.it wrote:

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

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 :)

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

Regards

----- Original Message -----
From: "Benny Prijono" bennylp@pjsip.org
To: "pjsip list" pjsip@lists.pjsip.org
Sent: Wednesday, March 19, 2008 6:45 PM
Subject: Re: [pjsip] Is possibile to have a on_rx_rtp callback with PJSUA?

On 3/19/08, Roland Klabunde roland.klabunde@freenet.de wrote:

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.

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

As I would do :)

But seems to be not that easy to capture the "start of ringing" in VoIP as
it was with ISDN...

Regards

Roland Klabunde wrote:

well, others suggestions to create the program that I need are all
welcome :)

Do you know any voip service that always send 180 Ringing exactly when
the remote phone is ringing? This can save me a lot of time !
Maybe I can open a new theard here in the list to ask if anyone knows a
service that always send the 180 ringing packet.

mmm, interesting, was is it more simple? Maybe I can consider to use an
ISDN with more fix costs if is really easy to implement.

Otherwise if there isn't another solution tell me: "good luck" :)

cheers
Davide