Re: [Live-devel] RTCP: Changing RR packets receptions interval.

> Is there any way, from the source side, to change the frequency of reception > of the RR packets. I presume that what you’re really asking is “How can you control the rate at which media receivers (e.g., RTSP clients) *send* RTCP RR packets?” Endpoints send RTCP packets in accordance with the

[Live-devel] RTCP: Changing RR packets receptions interval.

Hello, Is there any way, from the source side, to change the frequency of reception of the RR packets. I know I can change the transmission interval for sending the SR packets ( although it's getting dynamically calculated based on bandwidth, packet size, number of participants etc.) in rtc

Re: [Live-devel] RTCP RR reports do not reach to the server using UDP

> We are using Amazon cloud, Amazon EC2 g2xlarge Server Instance. IP multicast > is not supported on amazon server. But you should still be able to add a route for 224.0.0.0/4 for your ‘public’ IP address. Run (as root) route add 224.0.0.0/4 your-public-ip-address That would be the be

Re: [Live-devel] RTCP RR reports do not reach to the server using UDP

From: live-devel on behalf of Ross Finlayson Sent: Monday, November 9, 2015 3:35 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTCP RR reports do not reach to the server using UDP OK, you confused me by talking about ‘pri

Re: [Live-devel] RTCP RR reports do not reach to the server using UDP

OK, you confused me by talking about ‘private’ and ‘public’ addresses. That actually turns out to be irrelevant. The real issue is that your server is multi-homed (i.e., has more than one IP address), and is currently choosing (for LIVE555) an address that you don’t want. The best way to fix

Re: [Live-devel] RTCP RR reports do not reach to the server using UDP

s the way we did in above code. what should we do in this case? Best Regards Mohsin From: live-devel on behalf of Ross Finlayson Sent: Thursday, November 5, 2015 3:18 AM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel

Re: [Live-devel] RTCP RR reports do not reach to the server using UDP

It wasn’t totally clear from your message, but you seem to be implying that there is a NAT between your server and your client - with your server on the public Internet, and your client in a private IP address space (i.e., behind a NAT). In general, you *cannot* expect RTP/RTCP/RTSP streaming t

Re: [Live-devel] RTCP RR reports do not reach to the server using UDP

Best Regards Mohsin From: live-devel on behalf of Ross Finlayson Sent: Tuesday, November 3, 2015 11:28 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTCP RR reports do not reach to the server using UDP > But we are having problem with UDP

Re: [Live-devel] RTCP RR reports do not reach to the server using UDP

> But we are having problem with UDP (REQUEST_STREAMING_OVER_TCP false) . First > issue was that SR reports were sent by the server but were not reaching the > client. the wireshark on the server shows the ICMP destination unreacheable > (port unreachable) message from the client in response to

[Live-devel] RTCP RR reports do not reach to the server using UDP

We are streaming from the amazon aws EC2 server using live555 media streaming, and i am using testopenrtsp demo on the client side for testing. we want to use the RTSP RR packets for the checking of our client liveliness. Everything works well when we are recieving streams using RTP over TCP (

Re: [Live-devel] RTCP Bye packet length

, SO51 8BZ. Company Number : 03428325 From: live-devel [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ivan Roubícek Sent: 03 August 2015 08:09 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTCP Bye packet length Hi thank your for your repl. I understan

Re: [Live-devel] RTCP Bye packet length

Should you find a specific bug in our code, please let us know. Because you appear to not be using *our* software to receive/parse incoming RTCP packets, then we can’t help you on this mailing list. Sorry. Ross Finlayson Live Networks, Inc. http://www.live555.com/

Re: [Live-devel] RTCP Bye packet length

use Subject: Re: [Live-devel] RTCP Bye packet length You didn’t say specifically how your question refers to our software. (Remember that this mailing list is specifically for discussing the “LIVE555 Streaming Media” software.) If you are referring to RTCP packets that are sent by our software,

Re: [Live-devel] RTCP Bye packet length

You didn’t say specifically how your question refers to our software. (Remember that this mailing list is specifically for discussing the “LIVE555 Streaming Media” software.) If you are referring to RTCP packets that are sent by our software, then I believe you are mistaken. Our software sen

Re: [Live-devel] RTCP APP packet handling

> I can’t compile RTCP.cpp in Windows(because in Windows, it’s not have > snprintf.). Thanks for the report. I’ve now installed a new version (2015.02.12) of the code that should fix this. Ross Finlayson Live Networks, Inc. http://www.live555.com/ _

Re: [Live-devel] RTCP APP packet handling

Hi. I can't compile RTCP.cpp in Windows(because in Windows, it's not have snprintf.). When I used strncpy instead of snprintf, it's compiled. snprintf(nameBytes, 4, "%s", name); => strncpy(nameBytes, name, 4); thanks. ___ live-devel mailing lis

Re: [Live-devel] RTCP APP packet handling

Nick, I’ve now installed a new version (2015.02.10) of the “LIVE555 Streaming Media” code that includes support for sending/handling RTCP “APP” packets. I changed (simplified) your interface and implementation a little - in particular, there’s no longer “RTCPAppPacket” class. Because you’re t

Re: [Live-devel] RTCP unsynchronized

> I'm using the most current version of live555 on an embedded linux > platform to deliver an MPEGTS stream via RTSP. I am having issues where if > I use `testRTSPClient`, the debug prints for all payload/RTP packets have > the '!' at the end meaning they are not synchronized via RTCP. Where > shou

[Live-devel] RTCP unsynchronized

All, I'm using the most current version of live555 on an embedded linux platform to deliver an MPEGTS stream via RTSP. I am having issues where if I use `testRTSPClient`, the debug prints for all payload/RTP packets have the '!' at the end meaning they are not synchronized via RTCP. Where should I

Re: [Live-devel] RTCP APP packet handling

> Actually, that's true only if the "reuseFirstSource" parameter was set to > True. > If "reuseFirstSource" is False (its default value), then each client request > (to > "SETUP"/"PLAY" the stream) will create a new stream source (and a new > "StreamState" object). That makes sense now. Thanks f

Re: [Live-devel] RTCP APP packet handling

> What I’ve been unable to determine so far, is how the “RTCPInstance” > is selective about which clients it sends packets to. From what I > can see, sent packets are always delivered to all registered > clients (addresses in the groupsock). Since there is only a single > “StreamState” per subsess

Re: [Live-devel] RTCP APP packet handling

Hi Ross. > The additions to the "RTCPInstance" class (to assign a > RTCP "APP" packet handler, and send an "APP" packet) look > reasonable, and I'll probably add them to the library soon. That's good to know, thanks. > The changes to "OnDemandServerMediaSubsession", however, > don't look quite r

Re: [Live-devel] RTCP APP packet handling

Nick, The additions to the "RTCPInstance" class (to assign a RTCP "APP" packet handler, and send an "APP" packet) look reasonable, and I'll probably add them to the library soon. The changes to "OnDemandServerMediaSubsession", however, don't look quite right, and I'll need to think some more a

Re: [Live-devel] RTCP APP packet handling

From: live-devel [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: 16 December 2014 17:52 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTCP APP packet handling If we were to implement a way for a subclass of OnDemandMediaSubsessio

Re: [Live-devel] RTCP APP packet handling

> If we were to implement a way for a subclass of OnDemandMediaSubsession to > provide a custom handler for APP packets, in a similar way that handlers are > provided for the SR and RR packets, would you be likely to accept this as a > patch? Yes. Ross Finlayson Live Networks, Inc. http://www.

[Live-devel] RTCP APP packet handling

Hi Ross. We have a need to handle RTCP APP packets in our application. I notice from the Live555 RTCPInstance code that they are currently ignored. If we were to implement a way for a subclass of OnDemandMediaSubsession to provide a custom handler for APP packets, in a similar way that handlers

Re: [Live-devel] RTCP Sender Report Flooding

> Actually I try to have some support from a commercial linux in order to see I > it’s possible to fix this very old problem… Or, alternatively: 1/ Don't transmit to different multicast IP addresses using the same port number from the same host, or 2/ Make your multicast streams

Re: [Live-devel] RTCP Sender Report Flooding

devel-boun...@ns.live555.com] De la part de Ross Finlayson Envoyé : mercredi 3 juillet 2013 02:45 À : LIVE555 Streaming Media - development & use Objet : Re: [Live-devel] RTCP Sender Report Flooding We didn't understood deeply what brings the RTCP sender report flooding, but we has such a

Re: [Live-devel] RTCP Sender Report Flooding

> We didn’t understood deeply what brings the RTCP sender report flooding, but > we has such a behavior using multicast stream that share the same multicast > port. > Wireshark analysis show that an RTCP sender report is forwarded between 2 > RTSP servers without ends. > > We tried to understa

[Live-devel] RTCP Sender Report Flooding

Hi Ross, Thanks again for your support. We didn't understood deeply what brings the RTCP sender report flooding, but we has such a behavior using multicast stream that share the same multicast port. Wireshark analysis show that an RTCP sender report is forwarded between 2 RTSP serve

Re: [Live-devel] RTCP at wrong end in http tunneling

> I have tested one of my camera with live 555 for RTSP over http. > But i face some problem on RTCP side. > The "RR" packet was sent over GET side not on the POST side. > Also i have varified this on wireshardk. > Because of that problem server stops sending data after session time out. You have

[Live-devel] RTCP at wrong end in http tunneling

Dear sir, I have tested one of my camera with live 555 for RTSP over http. But i face some problem on RTCP side. The "RR" packet was sent over GET side not on the POST side. Also i have varified this on wireshardk. Because of that problem server stops sending data after session time out. So, can

Re: [Live-devel] RTCP Receiver Report information

> Is there a way to have an implementation where an application callback is > called every time an RTCP RR is received ? Yes, you can call "setRRHandler()" on the "RTCPInstance" object. See "liveMedia/include/RTCP.hh". Ross Finlayson Live Networks, Inc. http://www.live555.com/ __

Re: [Live-devel] RTCP Receiver Report information

Thanks Ross. In the proposed implementation the application has to periodically query the stats. Is there a way to have an implementation where an application callback is called every time an RTCP RR is received ? Regards, Sambhav On Apr 29, 2012, at 7:51 PM, Ross Finlayson wrote: >> How

Re: [Live-devel] RTCP Receiver Report information

> How can the OnDemand RTSP Server application get the RTCP Receiver Report > information ? Our RTSP servers (including "testOnDemandRTSPServer") automatically receive RTCP Receiver Reports from clients. Information obtained from these reports is stored in a database (class: "RTPTransmissionSt

[Live-devel] RTCP Receiver Report information

Hi, How can the OnDemand RTSP Server application get the RTCP Receiver Report information ? Regards, Sambhav ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

Re: [Live-devel] RTCP BYE from the RTSP Server

Thanks for the detailed Information. I am using live RTP source as stream source to a subclass of OnDemandServerMediaSubsession. In this case RTP when source stops sending data , the application gets a message upon which i want to close the session. On Mar 16, 2012, at 1:53 AM, Ross Finlays

Re: [Live-devel] RTCP BYE from the RTSP Server

> How to invoke a RTCP BYE message to the client from the RTSP Server > application (e.g testOnDemandRTSPServer) ? This will happen automatically when the stream ends - i.e., when the server reaches the end of the file that's being streamed. There is nothing that you need to do to get this; th

[Live-devel] RTCP BYE from the RTSP Server

Hi, How to invoke a RTCP BYE message to the client from the RTSP Server application (e.g testOnDemandRTSPServer) ? Regards, Sambhav ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

Re: [Live-devel] RTCP functions when using BasicUDPSource

Hi Ross, With your email, I found how to do it Thank you ! Ricardo On Wed, Feb 29, 2012 at 4:00 PM, Ross Finlayson wrote: > I would like to know what is the best way to get some of the RTCP info > when using UDP in the server side. > > Server side : we are using BasicUDPSource and StreamReplicato

Re: [Live-devel] RTCP functions when using BasicUDPSource

> I would like to know what is the best way to get some of the RTCP info when > using UDP in the server side. > > Server side : we are using BasicUDPSource and StreamReplicator to send > replicas towards the client apps. So, is your server's input data RTP/UDP, or raw-UDP? I.e., is your intent

[Live-devel] RTCP functions when using BasicUDPSource

On Thu, Jan 12, 2012 at 5:08 AM, Ross Finlayson wrote: > We have implemented a sender and a receiver for MPEG2TS using Livemedia. > > Do 'we' not have our own domain name? :-) > > Now "we" have a domaine :-) ! Hi Ross I would like to know what is the best way to get some of the RTCP info when u

Re: [Live-devel] RTCP Receiver Reports use in unicast sessions

> I tried again testMP3Streamer in multicast and now i receive back the RR. > Is there any special reason that works only in multicast (knowing RR is send > back in unicast to the original sender) ? I am trying to see how can it works > in unicast It works just fine with unicast if you use RTSP

Re: [Live-devel] RTCP Receiver Reports use in unicast sessions

Hi Ross Thank you for your response > > I tried with some other test as testMP3Streamer and testMP3receiver, and > I dont receive RTCP receive reports. > > > "testMP3Streamer" sends RTCP "SR" packets, and will receive RTCP "RR" > reports from any (multicast-connected receivers). "testMP3Receiver

Re: [Live-devel] RTCP Receiver Reports use

> We have implemented a sender and a receiver for MPEG2TS using Livemedia. > Do 'we' not have our own domain name? :-) > I found some old posts from 2004 talking about Receiver reports. > You realize, I hope, that 2004 was 8 years ago :-) We've had full support for RTCP "RR" and "SR" packets f

[Live-devel] RTCP Receiver Reports use

Hi Ross We have implemented a sender and a receiver for MPEG2TS using Livemedia. I found some old posts from 2004 talking about Receiver reports. "Note, however, that data from RTCP "RR" (Receiver Report) packets (i.e., coming from receivers back to the sender) are currently not processed at

Re: [Live-devel] RTCP bug in multicast?

I just read the rtcp.cpp and find a problem in multicast mode. In RTCPInstance::incomingReportHandler1(), from line 342 : [...] Here, the mechanism to prevent endless loop seems not perfect, checking the variables fHaveJustSentPacket and fLastPacketSentSize is not enough. Yes, you're

Re: [Live-devel] RTCP bug in multicast?

s? Something like a patch would be great. Best regards, Bruno Basilio Brisa Inovação e Tecnologia, S.A. > > -- > > Message: 5 > Date: Mon, 25 Apr 2011 14:10:32 +0800 > From: Mark > To: live-devel > Subject: [Live-devel] RTCP bug in multicast? >

[Live-devel] RTCP bug in multicast?

Hi Ross, I just read the rtcp.cpp and find a problem in multicast mode. In RTCPInstance::incomingReportHandler1(), from line 342 : // Ignore the packet if it was looped-back from ourself: if (RTCPgs()->wasLoopedBackFromUs(envir(), fromAddress)) { // However, we still want to handl

Re: [Live-devel] RTCP RR

Finlayson Sent: Monday, December 13, 2010 5:04 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTCP RR >I followed your suggestions (thank you) and indeed with openRTCP I >receive without problems the RTCP RRs. However, >VLC doesn't send any RTCP RR

Re: [Live-devel] RTCP RR

I followed your suggestions (thank you) and indeed with openRTCP I receive without problems the RTCP RRs. However, VLC doesn't send any RTCP RR (using exactly the same setup, stream). I made also sure that I am using the latest VLC 1.1.5. Could you please let me know how shall we proceed further?

Re: [Live-devel] RTCP RR

ll we proceed further? Thank you for your time, Monica -Original Message- From: live-devel-boun...@ns.live555.com [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: Friday, December 10, 2010 9:51 PM To: LIVE555 Streaming Media - development & use Subject: Re: [L

Re: [Live-devel] RTCP RR

I suggest that you begin by running "openRTSP" instead of VLC as your RTSP client. "openRTSP" should definitely be sending RTCP "RR" packets. If you see "openRTSP" sending "RR" packets, but VLC does not (and you're using the latest version of VLC), then let

Re: [Live-devel] RTCP RR

[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: Friday, December 10, 2010 12:47 AM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTCP RR >I ran into an issue where RTCP SRs are sent to my VLC client (playing >jpeg, h264 streams, UDP

Re: [Live-devel] RTCP RR

I ran into an issue where RTCP SRs are sent to my VLC client (playing jpeg, h264 streams, UDP transport, frequency 5 seconds but experimented with higher numbers as well), but no RTCP RRs are sent back. Could you please let me know when such situations occur? If your version of VLC is using the

[Live-devel] RTCP RR

Hello, I ran into an issue where RTCP SRs are sent to my VLC client (playing jpeg, h264 streams, UDP transport, frequency 5 seconds but experimented with higher numbers as well), but no RTCP RRs are sent back. Could you please let me know when such situations occur? Thank you for your

[Live-devel] RTCP RR handler

Hello, I want to implement an adaptive streaming algorithm in the livemedia server, based on the information sent in the RTCP reports and also in some additional RTCP APP reports. I studied a bit the livemedia code and I was thinking that the way to do this is to create a RRhandler which will

Re: [Live-devel] rtcp and NAT

Hi, Is RTSP server or client behind a NAT? For the case of RTSP client behind NAT, solutions are: 1. using RTSP application layer firewall proxy. 2. interleaving RTP/RTCP with RTSP, to transmit RTP/RTCP and RTSP via the same TCP connection. Shengye On Thu, Apr 8, 2010 at 6:44 PM, Alberto Alberic

[Live-devel] rtcp and NAT

Hi, is there a way to _force_  a client to send rtcp packets to another port than the one received in the rtsp handshaking? This would solve a lot of problems when the RTSP server is behind a NAT Otherwise, what could I do? thanks ___ live-de

Re: [Live-devel] rtcp is working wrong

i have implemented a streaming server (rtp and rtcp) and i have sometimes problem with rtcp. At system start up, sometimes rtcp don´t working and sometimes it´s working very fine. i made some statistic for 10 start up phases 2 times are wrong. I want to debung but i don´t know how. Before

[Live-devel] rtcp is working wrong

Hello all, i have implemented a streaming server (rtp and rtcp) and i have sometimes problem with rtcp. At system start up, sometimes rtcp don´t working and sometimes it´s working very fine. i made some statistic for 10 start up phases 2 times are wrong. I want to debung but i don´t know how

Re: [Live-devel] RTCP timeout

RTSPClient doesn't support the client timeout directly, so you have to wrap it (eg. subclass it) and schedule/reschedule your own task to do that. No, you don't need to do that. Instead, you can arrange for a 'liveness indicator' function (that you would write) to be called each time your clie

Re: [Live-devel] RTCP timeout

RTSPClient doesn't support the client timeout directly, so you have to wrap it (eg. subclass it) and schedule/reschedule your own task to do that. I can't remember what we use as the stimulus to reschedule, and don't have time to look right now. later, patbob On Thursday 23 April 2009 5:58

[Live-devel] RTCP timeout

Hi All,    I'm streaming a video using RTP over a wireless channel. If the server disappears, the client keeps sending RTCP report packets instead of checking if the server is still alive. Such behaviour is required by RFC 3550 (6.3.5 - Timing Out an SSRC) which states: "At occasional intervals, t

Re: [Live-devel] RTCP packets sent on wrong-end of HTTP tunnel

Op 27 jan 2009, om 14:05 heeft Ross Finlayson het volgende geschreven: this is just to let you know there's a small problem with RTCP reports when tunneling over HTTP. I noticed on my custom RTSP proxy that RTCP packets are sent on the GET side of the HTTP tunnel, i.e., the reply side Yes, y

Re: [Live-devel] RTCP packets sent on wrong-end of HTTP tunnel

Op 27 jan 2009, om 14:05 heeft Ross Finlayson het volgende geschreven: this is just to let you know there's a small problem with RTCP reports when tunneling over HTTP. I noticed on my custom RTSP proxy that RTCP packets are sent on the GET side of the HTTP tunnel, i.e., the reply side Ye

Re: [Live-devel] RTCP packets sent on wrong-end of HTTP tunnel

this is just to let you know there's a small problem with RTCP reports when tunneling over HTTP. I noticed on my custom RTSP proxy that RTCP packets are sent on the GET side of the HTTP tunnel, i.e., the reply side Yes, you're right - this is a bug. I'll try to fix it soon. -- Ross Finlayson

[Live-devel] RTCP packets sent on wrong-end of HTTP tunnel

Hi, this is just to let you know there's a small problem with RTCP reports when tunneling over HTTP. I noticed on my custom RTSP proxy that RTCP packets are sent on the GET side of the HTTP tunnel, i.e., the reply side. My HTTP handler breaks the connection because of the extra data tha

Re: [Live-devel] RTCP

I want send RTCP report to my server. So, I call myRTCTInstance->sendReport You don't need to do this. Just creating a "RTCPInstance" for your "RTPSource", and RTCP (Reception) reports will get sent automatically. You should never call "RTCPInstance::sendReport()" yourself. but It send U

[Live-devel] RTCP

Hi Ross,I want send RTCP report to my server.So, I call myRTCTInstance->sendReport but It send UDP packet to a multicast adress (224.2.2.2) and not at server unicast adress. Why?My server send me packet on port( audio) and port (video), so I store packets rtp in two files.Every for exam

Re: [Live-devel] RTCP

I need use RTCP SR and RR in my application. Remember that RTCP "SR" packets are sent only by RTP data transmitters, and RTCP "RR" packets are sent only by RTP data receivers. Live555 provide RTCPIstance class to do this, it's right? Yes. There are examples of use (best practice) in the

[Live-devel] RTCP

Hi Ross, I need use RTCP SR and RR in my application. Live555 provide RTCPIstance class to do this, it's right? There are examples of use (best practice) in the source code? Thanks in advance ___ live-devel mailing list live-devel@lists.live555.com htt

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

As I noted in my last message, you can use the existing RTSP client mechanism (using "openRTSP" as a model) to 1/ Detect when a stream has ended - using *either* RTCP "BYE* packets coming from the server, *or* (if the server keeps the stream open, without sending a "BYE") using the stream durati

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

On Fri, Oct 17, 2008 at 4:27 PM, Ross Finlayson <[EMAIL PROTECTED]> wrote: > >> >> In other words, once I do the above, what else >> >> do I have to do to clean up in rebuild again in preparation of >> >> streaming the next file? >> > >> > If you want to keep the existingoutput network connections

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

No, you can't (reliably) use RTCP "BYE" to trigger this, because the server will not always send a "BYE" when it reaches the stream. I meant to say, of course: No, you can't (reliably) use RTCP "BYE"s to trigger this, because the server will not always send a "BYE" when it reaches the end of

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

>> In other words, once I do the above, what else >> do I have to do to clean up in rebuild again in preparation of >> streaming the next file? > > If you want to keep the existingoutput network connections, and continue > streaming the next file, then you don't need to make *any* changes to th

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

On Fri, Oct 17, 2008 at 4:38 AM, Ross Finlayson <[EMAIL PROTECTED]> wrote: > >> In other words, once I do the above, what else >> do I have to do to clean up in rebuild again in preparation of >> streaming the next file? > > If you want to keep the existingoutput network connections, and continue

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

In other words, once I do the above, what else do I have to do to clean up in rebuild again in preparation of streaming the next file? If you want to keep the existingoutput network connections, and continue streaming the next file, then you don't need to make *any* changes to the existing

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

On Thu, Oct 16, 2008 at 6:17 AM, Ross Finlayson <[EMAIL PROTECTED]> wrote: > >> The behavior I expect is that at the end of the stream, when >> stopPlaying() is called on both sinks, live555 should send and RTCP >> Bye packet, causing MPlayer to exit. > > No, that's not what the code does. When th

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

At 04:17 PM 10/15/2008, you wrote: Hello, I am playing with the vobStreamer application, using MPlayer -rc2 to view the stream. The behavior I expect is that at the end of the stream, when stopPlaying() is called on both sinks, live555 should send and RTCP Bye packet, causing MPlayer to exit. B

Re: [Live-devel] RTCP Bye in PassiveServerMediaSession

The behavior I expect is that at the end of the stream, when stopPlaying() is called on both sinks, live555 should send and RTCP Bye packet, causing MPlayer to exit. No, that's not what the code does. When the code reaches the end of the input file, it reads/streams it again starting from th

[Live-devel] RTCP Bye in PassiveServerMediaSession

Hello, I am playing with the vobStreamer application, using MPlayer -rc2 to view the stream. The behavior I expect is that at the end of the stream, when stopPlaying() is called on both sinks, live555 should send and RTCP Bye packet, causing MPlayer to exit. But l have confirmed with wireshark th

Re: [Live-devel] RTCP SR clock sync diff

This is incorrect. The information in incoming RTCP "SR" packets is used to generate presentation times from incoming RTP packets' timestamps. These presentation times - like the RTP timestamps themselves - are (necessarily) based on the sender's clock (because that was the only clock availab

Re: [Live-devel] RTCP SR clock sync diff

> This is incorrect. The information in incoming RTCP "SR" packets is used > to generate presentation times from incoming RTP packets' timestamps. These > presentation times - like the RTP timestamps themselves - are (necessarily) > based on the sender's clock (because that was the only clock ava

Re: [Live-devel] RTCP SR clock sync diff

The RTCP SR NTP timestamp is only valid in direct comparison with a NTP timestamp from another stream of the same reference clock. So you can use it to synchronize the presentation times of multiple streams in the same session. You're not supposed to use it to directly generate a presentation t

Re: [Live-devel] RTCP SR clock sync diff

The RTCP SR NTP timestamp is only valid in direct comparison with a NTP timestamp from another stream of the same reference clock. So you can use it to synchronize the presentation times of multiple streams in the same session. You're not supposed to use it to directly generate a presentation time

Re: [Live-devel] RTCP SR clock sync diff

Sorry, but I won't be making any such change to the code. Once RTCP-generated presentation times become available, they should *always* be used. Note that the function "RTPSource:: hasBeenSynchronizedUsingRTCP()" can be used by a client to distinguish between 'guessed' initial presentation ti

[Live-devel] RTCP SR clock sync diff

I had a problem when using VLC to transcode from an AXIS camera into an RTP stream. In the RTCP Sender Report the AXIS camera is deriving its NTP time from the monotonic uptime instead of the wall clock. Then it converts it from unix epoch time into NTP time. The real issue is that VLC/live555 sta

Re: [Live-devel] RTCP auth bug

Mike, Thanks for the report. (BTW, you meant to say "RTSP auth", not "RTCP auth".) To properly debug this, I'll need the "rtsp://" URL of a stream that illustrates this problem, along with a valid username,password pair. (You can send this to me via private email if you wish.) -- Ross Finla

[Live-devel] RTCP auth bug

I just upgraded an AXIS camera's firmware to 4.47 (2008-03-13). The new firmware has gotten more strict with authorization causing a VLC 0.8.6h client to fail on an rtsp stream. I see the OPTIONS request succeed and then the DESCRIBE request fail w/o the authorization modifier. Then OPTIONS and

Re: [Live-devel] RTCP Synchronization information

Also, I was wondering if there was a searchable version of the live-devel archives? There are times when I would like to just search the messages first, before posting to the group It turns out that Google (and probably other search engines also) index our archives, so you can search them by a

Re: [Live-devel] RTCP Synchronization information

Ahh, I realized that after I posted the message - that's perfectly applicable for when I have an RTPSource feeding into my sink object, but for my audio situation, the SimpleRTPSource gets fed into an EndianSwap16 filter before going into my sink - in that case, its not possible to do that type

Re: [Live-devel] RTCP Synchronization information

Ahh, I realized that after I posted the message - that's perfectly applicable for when I have an RTPSource feeding into my sink object, but for my audio situation, the SimpleRTPSource gets fed into an EndianSwap16 filter before going into my sink - in that case, its not possible to do that typecast

Re: [Live-devel] RTCP Synchronization information

I know the variable fCurPacketHasBeenSynchronizedUsingRTCP in MultiFramedRTPSource.cpp has this information present in it, but how do I access it from the sink classes (i.e FileSink, etc) 1/ Call "MediaSink::source()" to get the sink's source object (as a (FramedSource*)). 2/ Cast this to a (R

[Live-devel] RTCP Synchronization information

Hello, I'm designing a playback application in Linux that plays back H.264 encoded video and uncompressed PCM audio to a screen in real-time as a RTP Client - I'm noticing that the presentation times I receive as a client often jump erratically until synchronization kicks in - since its i

Re: [Live-devel] RTCP and Live555 performance

Hi, I´m using the live555 to stream one image per seconde (reading from the files) and in the incomingHandler form the RTCPInstance i´m getting more client rtcp-app infos and i´m writting them in xml files every 5 secondes. The problem is after starting my application, the process performa

[Live-devel] RTCP and Live555 performance

Hi, I´m using the live555 to stream one image per seconde (reading from the files) and in the incomingHandler form the RTCPInstance i´m getting more client rtcp-app infos and i´m writting them in xml files every 5 secondes. The problem is after starting my application, the process performance i

[Live-devel] RTCP messages treatment

Hi all, I would like to know, in case it is implemented, how live555 analyzes and behaves regarding the RTCP information sent through this messages. That is, if it is implemented any change in the streaming session conditions that allows to improve the transmission quality when lost packets or hig

Re: [Live-devel] RTCP Extended Report Support in Live 555

>I need to implement the RTCP XR packet type. Is this support >already there in Live 555 ? No, not yet. Feel free to try implementing it. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ ___ live-devel mailing list live-devel@lists.live5

[Live-devel] RTCP Extended Report Support in Live 555

Hi, I need to implement the RTCP XR packet type. Is this support already there in Live 555 ? If not, do I just add another 'packet type' in the RTCP.hh file and give implementations to it ? Thanks in advance ! Sudhir ___ live-devel mailing list live-dev

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