Re: [Linphone-developers] oRTP pb during SIP refresh

[Simon Morlat]

Hi,

Yes it's a problem with particular gateways that do early-media or
session timers without media continuity.

Simon


Le jeudi 21 janvier 2010 à 10:43 +0100, Joerg Bergmann a écrit :
> Hi Simon,
> 
> As I understand: When I open a call with Linphone, the first second of 
> the stream is missing? When I do a call, my dialog partner says his 
> name or simply "Hello". This is done in about one second, and mostly 
> I'm missing that piece of stream... So I have problems with opening the 
> conversation, mostly. Is this due to that Linphone feature? I would
> call it a bug, commercial phones do not have that feature.
> 
> Joerg
> 
> Am Donnerstag, den 21.01.2010, 10:22 +0100 schrieb Frédérik Rouleau:
> > Hi Simon,
> > 
> > I cannot see why my patch is a problem. You are still locked to a
> > stream, it just enable to switch faster on a new stream when you ask
> > oRTP to.
> > Linphone has the same problem, at each refresh, you have a 1 second
> > silence (50 * 20ms). The gateway is a Cirpack, the same equipement
> > used by Free operator in France. Might be difficult to have any
> > upgrade on that side !!!
> > You also have this problem during ringing, when you answer the call,
> > the RTP stream is reset by the Cirpack, then you also have a 1 s
> > silence !
> > I am looking for a way to solve this issues. If you have better
> > suggestion, I am open to any proposal.
> > 
> > Regards,
> > 
> > Frédérik
> > 
> > 
> > 2010/1/19 Simon Morlat <address@hidden>
> >         Hi Frédérik,
> >         
> >         I understand this change solves your problem, but it is not
> >         really
> >         compatible with what this SSRC checking was done for.
> >         Initially the goal was to lock on the initial stream (and
> >         discard any
> >         others), and when one stream appears to be active (50 packet
> >         received
> >         without any other SSRC in between), we lock to this new
> >         stream.
> >         I think the gateway should not reinit the seq and timestamp
> >         numbers.
> >         
> >         RFC4028 was implemented in linphone quite recently. If you
> >         have both
> >         last svn/git of linphone and last svn of eXosip2, and have
> >         use_session_timers=1 in the [sip] section of the .linphonerc
> >         config
> >         file, they are enabled.
> >         
> >         Try it, this may workaround the gateway behaviour.
> >         
> >         Simon
> >         
> >         Le mardi 12 janvier 2010 à 16:03 +0100, Frédérik Rouleau a
> >         écrit :
> >         
> >         > Hello all,
> >         >
> >         > I am using oRTP and a SIP gateway which support SIP REFRESH
> >         (RFC
> >         > 4028). So at a fixed interval (600 s) I receive a SIP UPDATE
> >         but the
> >         > gateway also reinit the SSRC et seq number of the RTP flow.
> >         > The problem is that oRTP waits 1 second before it sends
> >         SSRC_CHANGED
> >         > message and during that ime I am loosing RTP paquets.
> >         Looking into the
> >         > code shows that there is a constant SSRC_CHANGED_THRESHOLD
> >         set at 50
> >         > setting this delay.
> >         >
> >         > Reducing the threshold might have some side effects but can
> >         we add a
> >         > kind of SSRC_change_pending state which set the flag
> >         > RTP_SESSION_RECV_SYNC on the first SSRC change paquet. for
> >         example the
> >         > rtp_parse code will look like:
> >         >
> >         >             if
> >         (session->inc_same_ssrc_count>SSRC_CHANGED_THRESHOLD){
> >         >
> >         >                 /* store the sender rtp address to do
> >         symmetric RTP */
> >         >                 if (!session->use_connect){
> >         >                     if (session->rtp.socket>0 &&
> >         > session->symmetric_rtp){
> >         >                         /* store the sender rtp address to
> >         do
> >         > symmetric RTP */
> >         >
> >         memcpy(&session->rtp.rem_addr,addr,addrlen);
> >         >                         session->rtp.rem_addrlen=addrlen;
> >         >                     }
> >         >                 }
> >         >                 session->rtp.rcv_last_ts = rtp->timestamp;
> >         >                 session->rcv.ssrc=rtp->ssrc;
> >         >
> >         rtp_signal_table_emit(&session->on_ssrc_changed);
> >         >             }else if (session->ssrc_change_pending){
> >         >                 session->ssrc_change_pending=FALSE;
> >         >                 rtp_session_set_flag(session,
> >         RTP_SESSION_RECV_SYNC);
> >         >             }else{
> >         >                 /*discard the packet*/
> >         >                 ortp_debug("Receiving packet with unknown
> >         ssrc.");
> >         >                 stats->bad++;
> >         >                 ortp_global_stats.bad++;
> >         >                 freemsg(mp);
> >         >                 return;
> >         >             }
> >         >
> >         > What do you think about it ? If you agree I can make changes
> >         and
> >         > propose a patch.
> >         >
> >         > Regards,
> >         >
> >         >
> >         
> >         > _______________________________________________
> >         > Linphone-developers mailing list
> >         > address@hidden
> >         > http://lists.nongnu.org/mailman/listinfo/linphone-developers
> >         
> >         
> > 
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