Mike wrote:
Hello,
I’ve developed a
custom
SIP application (using my own SIP library), and oRTP for the media
channel.
Everything seems
to work
great, other than some occasional DTMF issues (RFC2833 using the
rtp_session_send_dtmf()
function in oRTP).
If I run
wireshark (ethereal)
on the interface, and I send out a DTMF tone, normally I see five
packets
(three packets, then the third repeated two more times for
reliability)- which
correlates exactly with the code in telephonyevents.c. The maximum
duration
gets to 480.
But,
occasionally, I’ll
send a DTMF tone, and see hundreds of packets (the last test I did,
sent out
200 packets), and the duration keeps incrementing (the last test
incremented up
to 32160). It does also send the last packet three times as in the
first case
for reliability.
Because of the
long
duration, this obviously translates into a long tone on the receiving
end,
which is causing me some issues with some termination providers (some
deal with
it fine).
Does anybody have
any
thoughts on where these extra packets are coming from?
I’ve downloaded
the CVS
version of oRTP from the linphone distribution; I saw there was some
changes
related to the sequence number and timestamp, but it didn’t seem to
resolve this issue.
Any help would be
greatly
appreciated,
Mike
A Lot time ago i've fixed oRTP DTMF generating code, i don't remember
if my code finally made it's way to maing oRTP tree...
Here it goes:
=======================
/**
*rtp_session_create_telephone_event_packet:
address@hidden: a rtp session.
address@hidden: boolean to indicate if the marker bit should be set.
*
* Allocates a new rtp packet to be used to add named telephony
events. The application can use
* then rtp_session_add_telephone_event() to add named events to the
packet.
* Finally the packet has to be sent with
rtp_session_sendm_with_ts().
*
*Returns: a message block containing the rtp packet if successfull,
NULL if the rtp session
*cannot support telephony event (because the rtp profile it is bound
to does not include
*a telephony event payload type).
**/
mblk_t *rtp_session_create_telephone_event_packet(RtpSession
*session, int start)
{
mblk_t *mp;
rtp_header_t *rtp;
g_return_val_if_fail(session->snd.telephone_events_pt!=-1,NULL);
mp=allocb(RTP_FIXED_HEADER_SIZE+TELEPHONY_EVENTS_ALLOCATED_SIZE,BPRI_MED);
if (mp==NULL) return NULL;
rtp=(rtp_header_t*)mp->b_rptr;
rtp->version = 2;
rtp->markbit=start;
rtp->padbit = 0;
rtp->extbit = 0;
rtp->cc = 0;
rtp->ssrc = session->local.ssrc;
/* timestamp set later, when packet is sended */
/*seq number set later, when packet is sended */
/*set the payload type */
rtp->paytype=session->snd.telephone_events_pt;
/*copy the payload */
mp->b_wptr+=RTP_FIXED_HEADER_SIZE;
return mp;
}
/**
*rtp_session_send_dtmf:
address@hidden : a rtp session
address@hidden : a character meaning the dtmf (ex: '1', '#' , '9' ...)
address@hidden : the timestamp
*
* This functions creates telephony events packets for @dtmf and
sends them.
* It uses rtp_session_create_telephone_event_packet() and
* rtp_session_add_telephone_event() to create them and finally
* rtp_session_sendm_with_ts() to send them.
*
*Returns: 0 if successfull, -1 if the session cannot support
telephony events or if the dtmf
* given as argument is not valid.
**/
gint rtp_session_send_dtmf(RtpSession *session, gchar dtmf, guint32
userts)
{
return rtp_session_send_dtmf2(session, dtmf, userts, 480);
}
gint rtp_session_send_dtmf2(RtpSession *session, gchar dtmf, guint32
userts, int duration)
{
mblk_t *m1,*m2,*m3;
int tev_type;
int durationtier = duration/3;
/* create the first telephony event packet */
switch (dtmf){
case '1':
tev_type=TEV_DTMF_1;
break;
case '2':
tev_type=TEV_DTMF_2;
break;
case '3':
tev_type=TEV_DTMF_3;
break;
case '4':
tev_type=TEV_DTMF_4;
break;
case '5':
tev_type=TEV_DTMF_5;
break;
case '6':
tev_type=TEV_DTMF_6;
break;
case '7':
tev_type=TEV_DTMF_7;
break;
case '8':
tev_type=TEV_DTMF_8;
break;
case '9':
tev_type=TEV_DTMF_9;
break;
case '*':
tev_type=TEV_DTMF_STAR;
break;
case '0':
tev_type=TEV_DTMF_0;
break;
case '#':
tev_type=TEV_DTMF_POUND;
break;
case 'A':
case 'a':
tev_type=TEV_DTMF_A;
break;
case 'B':
case 'b':
tev_type=TEV_DTMF_B;
break;
case 'C':
case 'c':
tev_type=TEV_DTMF_C;
break;
case 'D':
case 'd':
tev_type=TEV_DTMF_A;
break;
case '!':
tev_type=TEV_FLASH;
break;
default:
g_warning("Bad dtmf: %c.",dtmf);
return -1;
}
m1=rtp_session_create_telephone_event_packet(session,1);
if (m1==NULL) return -1;
rtp_session_add_telephone_event(session,m1,tev_type,0,10,durationtier);
/* create a second packet */
m2=rtp_session_create_telephone_event_packet(session,0);
if (m2==NULL) return -1;
rtp_session_add_telephone_event(session,m2,tev_type,0,10,
durationtier+durationtier);
/* create a third and final packet */
m3=rtp_session_create_telephone_event_packet(session,0);
if (m3==NULL) return -1;
rtp_session_add_telephone_event(session,m3,tev_type,1,10,duration);
/* and now sends them */
rtp_session_sendm_with_ts(session,m1,userts);
rtp_session_sendm_with_ts(session,m2,userts+durationtier);
/* the last packet is sent three times in order to improve
reliability*/
m1=copymsg(m3);
m2=copymsg(m3);
/* NOTE: */
/* we need to copymsg() instead of dupmsg() because the buffers are
modified when
the packet is sended because of the host-to-network conversion of
timestamp,ssrc, csrc, and
seq number.
It could be avoided by making a copy of the buffer when sending
physically the packet, but
it add one more copy for every buffer.
Using iomapped socket, it is possible to avoid the user to kernel
copy.
*/
rtp_session_sendm_with_ts(session,m3,userts+durationtier+durationtier);
rtp_session_sendm_with_ts(session,m1,userts+durationtier+durationtier);
rtp_session_sendm_with_ts(session,m2,userts+durationtier+durationtier);
return 0;
}
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