Re: [Linphone-developers] DTMF issues with oRTP

[Simon Morlat]

Hi,

There is no reason why oRTP would send 200 packets when sending a dtmf.
I never saw such a behaviour ( oRTP is deployed in commercial telephony 
servers dealing thousand of channels simultaneously).
You should investigate in this two directions:
- a bug in your application: it probably also sends normal rtp packets ?
- a misinterpretation of wireshark: do the extra packets really come from the 
address and port used by your application ?

Simon

Le Monday 14 April 2008 19:20:55 Mike, vous avez écrit :
> Thanks Vadim,
>
>
>
> It looks like your changes did indeed make it into the tree- this code
> looks exactly (other than some of the obvious g* types and functions, and a
> few changes made after this, adjusting the timestamp and sequence number)
> like the code I'm using from the linphone distribution.
>
>
>
> This code looks fairly straight forward- I can't see any obvious reason why
> I would get the results I'm getting.
>
>
>
> I do also have a trace from wireshark that shows this output if that helps?
>
>
>
> Is there anywhere else in the code that deals with events? Or that might
> re-send packets for some reason?
>
>
>
> Thanks,
>
>
> Mike
>
>
>
> From: Vadim Lebedev [mailto:address@hidden
> Sent: Monday, April 14, 2008 6:32 AM
> To: Mike
> Cc: address@hidden
> Subject: Re: [Linphone-developers] DTMF issues with oRTP
>
>
>
> 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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>   _____
>
>
>
>
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