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  • H.264 over RTP - Identify SPS and PPS Frames

    - by Toby
    I have a raw H.264 Stream from an IP Camera packed in RTP frames. I want to get raw H.264 data into a file so I can convert it with ffmpeg. So when I want to write the data into my raw H.264 file I found out it has to look like this: 00 00 01 [SPS] 00 00 01 [PPS] 00 00 01 [NALByte] [PAYLOAD RTP Frame 1] // Payload always without the first 2 Bytes -> NAL [PAYLOAD RTP Frame 2] [... until PAYLOAD Frame with Mark Bit received] // From here its a new Video Frame 00 00 01 [NAL BYTE] [PAYLOAD RTP Frame 1] .... So I get the SPS and the PPS from the Session Description Protocol out of my preceding RTSP communication. Additionally the camera sends the SPS and the PPSin two single messages before starting with the video stream itself. So I capture the messages in this order: 1. Preceding RTSP Communication here ( including SDP with SPS and PPS ) 2. RTP Frame with Payload: 67 42 80 28 DA 01 40 16 C4 // This is the SPS 3. RTP Frame with Payload: 68 CE 3C 80 // This is the PPS 4. RTP Frame with Payload: ... // Video Data Then there come some Frames with Payload and at some point a RTP Frame with the Marker Bit = 1. This means ( if I got it right) that I have a complete video frame. Afer this I write the Prefix Sequence ( 00 00 01 ) and the NALfrom the payload again and go on with the same procedure. Now my camera sends me after every 8 complete Video Frames the SPS and the PPS again. ( Again in two RTP Frames, as seen in the example above ). I know that especially the PPS can change in between streaming but that's not the problem. My questions are now: 1. Do I need to write the SPS/PPS every 8th Video Frame? If my SPS and my PPS don't change it should be enough to have them written at the very beginning of my file and nothing more? 2. How to distinguish between SPS/PPS and normal RTP Frames? In my C++ Code which parses the transmitted data I need make a difference between the RTP Frames with normal Payload an the ones carrying the SPS/PPS. How can I distinguish them? Okay the SPS/PPS frames are usually way smaller, but that's not a save call to rely on. Because if I ignore them I need to know which data I can throw away, or if I need to write them I need to put the 00 00 01 Prefix in front of them. ? Or is it a fixed rule that they occur every 8th Video Frame?

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  • h264 RTP timestamp

    - by user269090
    Hi Guys, I have a confusion about the timestamp of h264 RTP packet. I know the wall clock rate of video is 90KHz which I defined in the SIP SDP. The frame rate of my encoder is not exactly 30 FPS, it is variable. It varies from 15 FPS to 30 FPS on the fly. So, I cannot use any fixed timestamp. Could any one tell me the timestamp of the following encoded packet. After 0 milisecond encoded RTP timestamp = 0 (Let the starting timestamp 0) After 50 milisecond encoded RTP timestamp = ? After 40 milisecond encoded RTP timestamp = ? After 33 milisecond encoded RTP timestamp = ? What is the formula when the encoded frame rate is variable? Thank you in advance.

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  • VLC RTP Streaming in FC12

    - by Matt D
    I'm trying to get VLC to work streaming RTP audio/video over my office network. The goal is multicast a/v streaming. In all test cases, we are streaming from VLC to VLC. I am able to stream from Windows to Windows, and from Fedora to Windows, but not from Windows to Fedora. Additionally, I am unable to receive a LOCAL stream from one instance of VLC to another, within Fedora. I don't see any reason why this would be. The buffer indicator (where the elapsed/total time is normally displayed) never shows any connectivity, so it would appear to be a network problem, but since I am able to stream from Fedora to Windows (same IP, same port) I thought it would be something else. Does anyone know of a solution to this issue?

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  • "RFC 2833 RTP Event" Consecutive Events and the E "End" Bit

    - by brian_d
    Hello, I can send out a RFC 2833 dtmf event as outlined at http://www.ietf.org/rfc/rfc2833.txt When I do set the E "End" bit, but leave it as 0, I get the following behaviour: If for example keys 7874556332111111145855885#3 were pressed, then ALL events would be sent and show up in a program like wireshark, however only 87456321458585#3 would sound. So the first key (which I figure could be a separate issue) and any repeats of an event (ie 11111) are failing to sound. In section 3.9, figure 2 of the above linked document, they give a 911 example. Here all but the last event have the E bit set. When I set the bit for all numbers, I never get an event to sound. I have thought of a couple possible thing but do not know if they are the reason: 1) figure 2 shows payload types of 96 and 97 sent. I have not nor know how to exactly. In section 3.8, codes 96 and 97 are described as "the dynamic payload types 96 and 97 have been assigned for the redundancy mechanism and the telephone event payload respectively" 2) In section 3.5, "E:", "A sender MAY delay setting the end bit until retransmitting the last packet for a tone, rather than on its first transmission" Does anyone have an idea of how to actually do this? I have also fiddled around with timestamp intervals and the RTP marker. Any help is greatly appreciated. Here is a sample wireshark event capture of the relevant areas: 6590 31.159045000 xx.x.x.xxx --.--.---.-- RTP EVENT Payload type=RTP Event, DTMF Pound # (end) Real-Time Transport Protocol Stream setup by SDP (frame 6225) Setup frame: 6225 Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: telephone-event (101) Sequence number: 0 Extended sequence number: 65536 Timestamp: 0 Synchronization Source identifier: 0x15f27104 (368210180) RFC 2833 RTP Event Event ID: DTMF Pound # (11) 1... .... = End of Event: True .0.. .... = Reserved: False ..00 0000 = Volume: 0 Event Duration: 2048

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  • process of connecting RTP with SIP via SDP & land lines

    - by TacB0sS
    Hello to everyone, I have a problem with starting a media session and to combine it with my SIP client. I've designed a recursive SIP client that reuse the same request template to send the next requests to server, according to the acceptable sequences noted in the RFC's, and examples that I read. as far as I could tell the SIP part is working fine registers to server invites, and authenticates. I didn't complete any calls to clients yet because of the content header needs to be filled up (which I didn't yet so I get a 503 from the server which is OK I guess). for a long time I didn't know where to start with the media session, and slowly learned how to use the JMF and I've constructed an object that handles RTP transmitting, now I'm standing at the cross road, on the one hand I have my SIP signaling but it needs the SDP content header to complete the invite, and on the other I have the RTP which is knows how to p2p. For me to complete my design I require your help with the following questions: Is there an easy//a simple//an implemented way to convert the Audio/Video format from the JMF into SDP media headers? or even a generator that I would input all the parameters for the content header, and it would generate a content header fast, or do I have to implement this myself? Once I've finished constructing the SDK and the SIP is up and running and I get an OK response from the server (after ringing and all), how do I start the media session? do I connect p2p according to caller details I send in the SIP invite? If 2 is correct, then how does a connection to land lines would be? does land lines knows that once they send an OK back to server they listen/start RTP session on a specific port? Or did I get everything wrong? :-/ I really appreciate any help I could I get, I looked every where for answers but they are not clear, they ignore question 2 as if it was an obvious thing, but for me it just isn't. Thank in advance, Adam Zehavi.

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  • local RTP port unreachable when using mjsip/jmf

    - by brian_d
    Hello, I create a sip session with mjsip to an external voip provider. Then I transmit a test wav file over rtp to the provider using RtpManager. The program runs with no errors and I answer the sip call. However, no audio is transmitted. When I diagnose the network traffic with wireshark, I see a bunch of RTP traffic from my localhost (behind some kind of nat) to the voip provider and nothing back. After a while I get the ICMP error "Destination unreachable (Port unreachable)" from the provider to my localhost. The software linphone works using the same localhost and voip provider - though it is using a different sip stack. Any suggestions? Thanks

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  • does android natively support RTP and/or SCTP?

    - by user299988
    Hi, I am developing a walkie-talkie application for Android, and would like to know whether RTP and/or SCTP is natively supported in the latest version of android SDK. From whatever I have dug-up so far, the answer is no. It seems that support for SCTP is coming in JDK 7. But then, I am not sure how this will work with my android developemnt. Could you please shed some light over this? Thanks,

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  • RTP Client Application on Android Mobile Device

    - by devang
    Hey folks,i am developing a RTP client on an Android device which can play streaming videos from a server. I am confused regarding how should i start about? i am thinking of developing a web app, using HTML,CSS and Javascript, which can later be wrapped in Android.is this approach correct? does javascript support real time media player? please guide me, i am a fresher and completely clueless..:(

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  • audio and video data in RTP

    - by Banana
    Suppose a user wants to transmit both audio and video to another user, whose formats are AMR for audio and H.264 for video. Does the user have to transmit audio and video packets always separately? Meaning that it is not possible to mix audio and video within the same RTP packed, is that correct? If this is true I guess the RTP protocol will need to know the SSRC of both audio and video to be able to check the sync of the two streams.

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  • RTP H.264 save and replay

    - by user301810
    We are interested in saving a H.264 stream and replaying it. Is there any one who experience saving h.264 using winpcap and replaying it. We were able to save H.263 and replay, but same logic does not work for H.264. We also tried rtpdump tool to save H264 stream, but we were unable to replay it in that format? thanks in advance

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  • How to use RTPSocket to send RTP packets

    - by Afro Genius
    Hi there, am relatively new to JMF but have gone through the documents and have a sufficient understanding of how it works. That been said am having some trouble implementing a the server side for RTPSockets. After looking at their illustrations and example. I am still abit confused. Am I to develop a datasource and also datasink classes to handle the transfer? What am trying to do is stream data from my application to the underlying network and receive it back through another application. I have and understand receiving but just can't get my head around the steps involved for sending. Any help would be most appreciated.

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  • help bonding streaming rtp 3g

    - by enrique
    first sorry for contact me here. Recuro to you after reading all the material I found about it and so it does not get set. My question is: I can configure load balancing in any way out? I have a hub with 3 USB 3G modems, I got the 3 simultaneously connect with an upload speed of about 500kb in each approx. and a dynamic ip each. I do a unicast streaming with vlc rtp with a bandwidth of 1.5mb. Bone the sum of the three modems. I was searching on ifenslave, iproute. Then I found a draft vlc MultiCat. I understood that this could end, but configure it only moves a card. If I can help extend the information willingly. From now eternally grateful.

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  • Is there simple way to play an rtp video/audio stream in WPF?

    - by Robin
    I need to create a WPF control that will play an rtp stream with the requirement that the latency needs to be as low as possible. I've looked at the following two projects: http://vlcdotnet.codeplex.com/ http://wpfmediakit.codeplex.com/ As far as I know, I can't use VLC because we're shipping a commercial application with a more restrictive license than GPL (i.e. we can't ship our source). Wpf media kit is nice, but I can't seem to find a good/free rtp directshow source filter and I wanted to ask if there is a simpler solution out there that I'm missing before I jump into writing my own. Any ideas?

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  • MPEG2-TS streaming: UDP or RTP?

    - by Juan Jose Polanco Arias
    Hello I'm working on an IPTV streaming server in Linux (Ubuntu Server 12.04 LTS) that has a DVB-S/S2 card to obtain satellite channels. Then with MuMuDVB I map all channels in the transponder to a multicast group, for multicast transmission. Now for the MuMuDVB software I can either use UDP for transmission or I can add the RTP header. I was wondering what would be the most convenient for MPEG2-TS because I've heard that RTP is used primarily for MPEG4, but It's also said that RTP can be used for MPEG2-TS. Thanks for your help.

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  • 1 VoIP Conversation but 2 RTP Streams?

    - by pepito
    I'm testing a VoIP system based on OpenSIPS. It has no RTPproxy, so calls do not pass through OpenSIPS. I tried to make a call between two smartphones, and it succeeded. I also turned on Wireshark, and got this result. Is that mean that voice call from 1st phone to 2nd phone went through 1st RTP stream and voice call from 2nd phone to 1st phone went through 2nd RTP stream? Why couldn't it only used one RTP stream? It could just go back and forth :)

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  • How to force client to switch RTP transport from UDP to TCP?

    - by Cipi
    If the client wants to watch a stream that is on my RTSP server, it first tries to setup a stream through the UDP protocol. How can I tell it that my server only supports RTP/AVP/TCP and that it should switch transports? I want to terminate the UDP support on my server, but all the clients first try to SETUP the session over UDP, and later they do so over TCP... and I want to switch them to TCP as soon as possible in RTSP protocol. How can I do that?

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  • How do I determine if a packet is RTP/RTCP?

    - by Chris Holmes
    I am using SharpPCap which is built on WinPCap to capture UDP traffic. My end goal is to capture the audio data from H.323 and save those phone conversations as WAV files. But first thing is first - I need to figure out what my UDP packets are crossing the NIC. SharpPCap provides a UdpPacket class that gives me access to the PayloadData of the message. But I am unsure what do with this data. It's a Byte[] array and I don't know how to go about determining if it's an RTP or RTCP packet. I've Googled this topic but there isn't much out there. Any help is appreciated.

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  • How to seek to a specific time in a RTP stream?

    - by Cipi
    I am streaming a prerecorded H264 video that has the following structure: [I] [x] [x] [x] [I] [x] [x] [x] [I]... In between the IDR (I-s in my structure) I have 32 (only 3 presented here) other frames (all other stuff that is not IDR like SEI, SPS, PPS... X-es) Now, let assume that the timing of my frames is such: TIME: 1 2 3 4 5 6 7 8 9 FRAME: [I] [x] [x] [x] [I] [x] [x] [x] [I]... Now i want to seek to the time 4. If I seek to that frame, and send it, the picture gets messed up because the decoder needs a IDR to decode it properly, so I resorted to finding the appropriate IDR (in this case one with the time 1) and sending it as the frame with the time 4. So now the picture is decoded properly, all is well... but... If my GOV is 32, and I need to send the non IDR frame that has the index 31, and if the time span between it and the corresponding IDR is 3 seconds, I actually get 3 seconds earlier then the time I want. Now, this is not precise, because I cannot seek to the half of the GOV time span. Also, I cant set smaller GOV, so I want other ideas... My other idea was to send the last known IDR, and then send all other non IDR frames that come before the one I want, only I would set for all of them RTP-TIME to be the same as the corresponding IDR. In this case the picture gets decoded perfectly, but now in the above case, 3 seconds that follow non IDR frame with the wanted time get fast paced in the decoder/player (there is no instantaneous seek)... Any ideas? Or I can only seek to IDR-s and not the frames in between?

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  • Checking rtp stream audio quality.

    - by chills42
    We are working in a test environment and need to monitor the audio quality of an rtp stream that is being captured using tshark. Right now we are able to capture the audio and access the file through wireshark, but we would like to find a way to save the audio to a .wav file (or similar) via the command line. Does anyone know of a tool that can do this?

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  • How to debug packet loss ?

    - by Gene Vincent
    I wrote a C++ application (running on Linux) that serves an RTP stream of about 400 kbps. To most destinations this works fine, but some destinations expericence packet loss. The problematic destinations seem to have a slower connection in common, but it should be plenty fast enough for the stream I'm sending. Since these destinations are able to receive similar RTP streams for other applications without packet loss, my application might be at fault. I already verified a few things: - in a tcpdump, I see all RTP packets going out on the sending machine - there is a UDP send buffer in place (I tried sizes between 64KB and 300KB) - the RTP packets mostly stay below 1400 bytes to avoid fragmentation What can a sending application do to minimize the possibility of packet loss and what would be the best way to debug such a situation ?

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