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• RTP session with Gstreamer

  - by Walidix

  I'm newbie with Gstreamer and I'm trying to use it in order to make a RTP session I can make a Gstrtpbin sender and a Gstrtpbin receiver separately but I can not make the same Gstrtpbin as sender and receiver in the same time. My question: Is it possible to do it ??? If it is, I will be thankful for a simple example with the C language.

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• iphone api sip rtp srtp zrtp

  - by user345974

  hi,I would like to ask if there are , or there is one of this Api:sip rtp srtp zrtp for iphone , if there is a way for to use them in the iphone simulator , and what is the default gataway in iphone .

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• How does one capture H.323 voice traffic on a VOIP network?

  - by Chris Holmes

  What I am trying to do is capture the WAV data of a phone conversation on a VOIP network using SharpPCap/PCap.Net. We are using the H.323 recommendation and my understanding is that voice data is located in the RTP packets. However, there is no way to heuristically determine if a UDP packet is a RTP packet, so we have to do more work before we can capture the data. The H.323 recommendation apparently uses a lot of traffic on specific TCP ports to negotiate the call before the WAV data is sent via RTP. However, I am having very little luck determining what data is actually sent on those TCP ports, when it is sent, what the packets look like, how to handle it, etc. If anyone has any information on how to go about this I'd really appreciate it. My Google-Fu seems to be failing me on this one.

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• GUI Control For Audio Presentation

  - by Boris

  I need GUI control for audio file presentation. The language is not very important but it should run on windows platform. I should be able to :- load the file play the sound put and move markers across the audio bar. it would be nice if it can load itself from RTP wireshark captures (and not wav files). An example may be seen in audacity (may be someone even had an experience extracting it from there). Writing nyquist scripts in audacity is not a good option because I have to operate on RTP captures and not on raw sound samples. Another example of such control is wireshark RTP analyzer. Any advise?

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• H.323 to RTSP gateway?

  - by davr

  Is there such a thing as a H.323 to RTSP gateway? Am I searching for the wrong terms? This site seems to imply that such a thing should already exists, but I cannot find anything at all. My end goal is to connect a Flash applet (via RTMP) on one end to a video conference (which uses H.323) on the other end. I have a RTMP<--RTSP/RTP gateway, so a RTSP<--H.323 gateway would allow this solution to work.

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• how to continuously send data without blocking?

  - by Donal Rafferty

  I am trying to send rtp audio data from my Android application. I currently can send 1 RTP packet with the code below and I also have another class that extends Thread that listens to and receives RTP packets. My question is how do I continuously send my updated buffer through the packet payload without blocking the receiving thread? public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); Log.d("BUFFERSIZE","Buffer size = " + buffersize); arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); short[] readBuffer = new short[80]; byte[] buffer = new byte[160]; arec.startRecording(); while(arec.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING){ int frames = arec.read(readBuffer, 0, 80); @SuppressWarnings("unused") int lenghtInBytes = codec.encode(readBuffer, 0, buffer, frames); RtpPacket rtpPacket = new RtpPacket(); rtpPacket.setV(2); rtpPacket.setX(0); rtpPacket.setM(0); rtpPacket.setPT(0); rtpPacket.setSSRC(123342345); rtpPacket.setPayload(buffer, 160); try { rtpSession2.sendRtpPacket(rtpPacket); } catch (UnknownHostException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (RtpException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (IOException e) { // TODO Auto-generated catch block e.printStackTrace(); } } } So when I send on one device and receive on another I get decent audio, but when I send and receive on both I get broken sound like its taking turns to send and receive audio. I have a feeling it could be to do with the while loop? it could be looping around in there and not letting anything else run?

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• Why does use of H264 in sender/receiver pipelines introduce just HUGE delay?

  - by Serguey Zefirov

  When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong. I took pipelines from gstrtpbin documentation page and slightly modified them to use Speex: This is sender pipeline: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 Receiver pipeline: !/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false Those pipelines, a combination of H263 and Speex, work fine enough. I snap my fingers near camera and micropohne and then I see movement and hear sound at the same time. Then I changed pipelines to use H264 along the video path. The sender becomes: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! x264enc bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 And receiver becomes: #!/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false This is what happen under Ubuntu 10.04. I didn't noticed such huge delays on Ubuntu 9.04 - the delays there was in range 2-3 seconds, AFAIR.

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• Android stream to Wowza

  - by Curtis Kiu

  I feel very confused about Android streaming to wowza. I am doing a video conference using rtmp cross-platform, but Android doesn't eat RTMP. Therefore I need to find another way to do it. Upstreaming I found a new open-source app called spydroid-ipcamera. It is using rtp, sending udp packets to computer, and opens it in vlc using the following sdp v=0 s=Unnamed m=video 5006 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1;profile-level-id=420016;sprop-parameter-sets=Z0IAFukBQHsg,aM4BDyA=; But it can't work. Then I follow wowza tutorial and stream to it and then play again in VLC. That works! I wrote it in http://code.google.com/p/spydroid-ipcamera/issues/detail?id=2 However when I want to add audio in the packet, it fails to work. I change to code in http://code.google.com/p/spydroid-ipcamera/source/browse/trunk/src/net/mkp/spydroid/CameraStreamer.java mr.setAudioSource(MediaRecorder.AudioSource.MIC); mr.setVideoSource(MediaRecorder.VideoSource.CAMERA); mr.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4); mr.setVideoFrameRate(20); mr.setVideoSize(640, 480); mr.setAudioEncoder(MediaRecorder.AudioEncoder.AAC); mr.setVideoEncoder(MediaRecorder.VideoEncoder.H264); mr.setPreviewDisplay(holder.getSurface()); Then I thought that the problem should be in sdp, but I don't know how to due with sdp. I am streaming H.264/AAC with Mp4 Second I don't understand sdp. So how can I make video conference upstreaming part using this apps. Android ----(UDP Port:5006)----> PC (SDP file) and then Wowza read the SDP file ------> VLC I think in this way the system cannot handle more than 1 client. sdp can only hold 1 port, any idea or actually it wont' work? Also Wowza need to set the stream before we stream it, so does it mean that I should not follow this way to do it? Sorry my English is poor, I hope you guys understand.

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• JMF RTPManager transmitting side

  - by TacB0sS

  I was wondering please, the RTP manager in the JMF can perform as a uni-cast,multi-cast, uni-multi-cast, if the session is multi cast the you add the local address to the target list, why is that? what is the logic and effect behind this? thanks for your help, Adam.

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• raw h.264 packet capture and playing in VLC

  - by MAC

  Hi, I am capturing packets off the network from a video conference HDX. The video is sent in RTP and is encoded in H264. I am trying to capture these packets and generate a video file. I wrote raw H264 data from the packets to disk and i am trying to play it in VLC. VLC just shows a green box. Am i being too naive in my approach with data writing or should am I wrong in assuming that VLC should play this file? Anyone have any experience in such things?

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• Using VLC as RTSP server

  - by StackedCrooked

  I'm trying to figure out how to use the server capabilities of VLC. More specifically, how to export an SDP file when RTP streaming. In chapter 4 in the section related to RTP Streaming examples for server and client are given: vlc -vvv input_stream --sout '#rtp{dst=192.168.0.12,port=1234,sdp=rtsp://server.example.org:8080/test.sdp}' vlc rtsp://server.example.org:8080/test.sdp It's not very clear to me how to make it actually work. I have tried these two commands for server and client using two cmd instances: vlc -I rc screen:// --sout=#rtp{dst=127.0.0.1,port=4444,sdp=rtsp://localhost:8080/test.sdp} vlc -I rc rtsp://localhost:8080/test.sdp Invoking the second command causes the first one to crash. The second command shows the error message "could not connect to localhost:8080".

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• Asterisk terminating outbound call when picked up, sends 'BYE' message

  - by vo

  I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway the setup does not appear to have issues registering and setting up the call, RTP packets go both ways and you can hear the ringing from the other side. However it appears that when the call is picked up or thereabouts, the incoming RTP packets cease. Upon closer inspection with Wireshark, there are these particular packets that seem to be the cause: trunk->asterisk SIP/SD Status: 200 OK, with session description asterisk->trunk SIP Request: ACK sip:<phone>@trunk:6889 asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 [..about a dozzen RTP packets in/outbound..] trunk->asterisk SIP Status: 200 OK, CSeq: 104 Bye [..outbound RTP continues, phone is silent..] Then the inbound RTP packets cease, however the asterisk logs dont show any activity at this point. The last entry reads 'SIP/ is answered SIP/'. Then when you hangup the extension, you get asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 trunk->asterisk SIP Status: 481 Call Leg/Transaction does not exist My trunk peer settings in FreePBX are: username=<user> fromuser=<user> canreinvite=no type=friend secret=<pass> qualify=no [qualify yes produces 401/forbidden messages] nat=yes insecure=very host=<sip trunk gateway> fromdomain=<sip trunk gateway> disallow=all context=from-pstn allow=ulaw dtmfmode=inband Under sip_general_custom.conf i have stunaddr=stun.xten.com externrefresh=120 localnet=192.168.1.1/255.255.255.0 nat=yes Whats causing Asterisk to prematurely end the call and still think the call is in progress? I have no idea where to look next.

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• VLC desktop streaming

  - by StackedCrooked

  Edit I stopped using VLC and switched to GMax FLV Encoder. It does a much better job IMO. Original post I am sending my desktop (screen) as an H264 video stream to another machine that saves it to a file using the follwoing command lines: Sender of the stream: vlc -I dummy --sout='#transcode{vcodec=h264,vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' Receiver of the stream: vlc -I rc rtp://@:4444 --sout='#std{access=file,mux=ps,dst=/home/user/output.mp4}' --ipv4 This works, but there are a few issues: The file is not playable with most players. VLC is able to playback the file but with some weirdness: = it takes about 10 seconds before the playback actually begins. = seeking doesn't work. Can someone point me in the right direction on how to fix these issues? EDIT: I made a little progress. The initial delay in playback is because the player is waiting for a keyframe. By forcing the sender of the stream to create a new key-frame every 4 seconds I could decrease the delay: :screen-fps=10 --sout='#transcode{vcodec=h264,venc=x264{keyint=40},vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' The seeking problem is not solved however, but I understand it a little better. The RTP stream is saved as a file in its original streaming format, which is normally not playable as a regular video file. VLC manages to play this file, but most other players don't. So I need to convert it to a regular video file. I am currently investigating whether I can do this with ffmpeg if I provide it with an SDP file for the recorded stream. All help is welcome!

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• Filling a byte array in Java

  - by Corleone

  Hey all! For part of a project I'm working on I am implementing a RTPpacket where I have to fill the header array of byte with RTP header fields. //size of the RTP header: static int HEADER_SIZE = 12; // bytes //Fields that compose the RTP header public int Version; // 2 bits public int Padding; // 1 bit public int Extension; // 1 bit public int CC; // 4 bits public int Marker; // 1 bit public int PayloadType; // 7 bits public int SequenceNumber; // 16 bits public int TimeStamp; // 32 bits public int Ssrc; // 32 bits //Bitstream of the RTP header public byte[] header = new byte[ HEADER_SIZE ]; This was my approach: /* * bits 0-1: Version * bit 2: Padding * bit 3: Extension * bits 4-7: CC */ header[0] = new Integer( (Version << 6)|(Padding << 5)|(Extension << 6)|CC ).byteValue(); /* * bit 0: Marker * bits 1-7: PayloadType */ header[1] = new Integer( (Marker << 7)|PayloadType ).byteValue(); /* SequenceNumber takes 2 bytes = 16 bits */ header[2] = new Integer( SequenceNumber >> 8 ).byteValue(); header[3] = new Integer( SequenceNumber ).byteValue(); /* TimeStamp takes 4 bytes = 32 bits */ for ( int i = 0; i < 4; i++ ) header[7-i] = new Integer( TimeStamp >> (8*i) ).byteValue(); /* Ssrc takes 4 bytes = 32 bits */ for ( int i = 0; i < 4; i++ ) header[11-i] = new Integer( Ssrc >> (8*i) ).byteValue(); Any other, maybe 'better' ways to do this?

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• Browser Based Streaming Video/Audio (not progressive download)

  - by Josh

  Hello, I am trying to understand conceptually the best way to deliver real streaming audio and video content. I would want it to be consumed with a web browser, utilizing the least amount of proprietary technology. I wouldn't be serving static files and using progressive download, this would be real audio streams being captured live. How does one broadcast a stream that will be reasonably in sync with the source? What kind of protocol is suitable? Edit: In research I've found that there are a few protocols: RTSP, HTTP Streaming, RTMP, and RTP. HTTP streaming is somewhat unsuitable if you are streaming a live performance/communication of some kind because it relies on TCP (as its HTTP based) and you don't lose packets. In a low bandwidth situation, the client can get significantly behind in playback. ref RTMP is a proprietary technology, requiring flash media server. Crap on that. The reason I looked at flash is because they are extremely flexible as far as user experience goes. SoundManager2 provides an excellent javascript interface for playing media with flash. This is what I would look for in a client application. RTSP/RTP is what Microsoft switched to using, deprecating their MMS protocol. RTSP is the control protocol. Its similar to HTTP with a few distinct difference -- server can also talk to the client, and there are additional commands, like PAUSE. Its also a stateful protocol, which is maintained with a session id. RTP is the protocol for delivering the payload (encoded audio or video). There are a few open sourced projects, one of them being supported by apple here. It seems like this might do what I want it to, and it looks like quite a few players support it. It sounds like it would be suitable for a "live" broadcast from this page here. Thanks, Josh

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• Having trouble Getting "RTSP over HTTP"

  - by Muhammad Adeel Zahid

  There is an axis camera that is connected to our site (camba.tv) through axis one click connection component (which acts as proxy). We can communicate with this camera only through http by setting the proxy to our OCCC server's address. If we want to get RTSP streams (h.264) we are only left with "RTSP over HTTP" option. For this I have followed axis VAPIX 3 documentation section 3.3. I issue requests through fiddler but don't get any response. But when i put the URL (axrtsphttp://1.00408CBEA38B/axis-media/media.amp) in windows media player (with proxy set to OCCC server 212.78.237.156:3128) the player is able to get RTSP stream over HTTP after logging in. I have created a request trace of communication between camera and windows media player through wireshark and the request that brings the stream looks like http://1.00408cbea38b/axis-media/media.amp HTTP/1.1 x-sessioncookie: 619 User-Agent: Axis AMC Host: 1.00408CBEA38B Proxy-Connection: Keep-Alive Pragma: no-cache Authorization: Digest username="root",realm="AXIS_00408CBEA38B",nonce="000a8b40Y0100409c13ac7e6cceb069289041d8feb1691",uri="/axis-media/media.amp",cnonce="9946e2582bd590418c9b70e1b17956c7",nc=00000001,response="f3cab86fc84bfe33719675848e7fdc0a",qop="auth" HTTP/1.0 200 OK Content-Type: application/x-rtsp-tunnelled Date: Tue, 02 Nov 2010 11:45:23 GMT RTSP/1.0 200 OK CSeq: 1 Content-Type: application/sdp Content-Base: rtsp://1.00408CBEA38B/axis-media/media.amp/ Date: Tue, 02 Nov 2010 11:45:23 GMT Content-Length: 410 v=0 o=- 1288698323798001 1288698323798001 IN IP4 1.00408CBEA38B s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50000 t=0 0 a=control:* a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:30.0 a=transform:1,0,0;0,1,0;0,0,1 a=control:trackID=1 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeNQFAe2AtwEBAaQeJEV,aM48gA== RTSP/1.0 200 OK CSeq: 2 Session: 3F4763D8; timeout=60 Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=060922C6;mode="PLAY" Date: Tue, 02 Nov 2010 11:45:24 GMT RTSP/1.0 200 OK CSeq: 3 Session: 3F4763D8 Range: npt=0- RTP-Info: url=rtsp://1.00408CBEA38B/axis-media/media.amp/trackID=1;seq=7392;rtptime=4190934902 Date: Tue, 02 Nov 2010 11:45:24 GMT [Binary Stream Content] But when i copy this request to fiddler, I only get 200 status code with content-type set to application/x-rtsp-tunneled and there is no stream data. The only thing i do different with stream is to use Basic in authorization header instead of Digest and I do not get 401 (Un authorized) status code. Can anyone explain what's happening here? How can I write request sequences to get stream in fiddler? If it is needed, I can upload the wireshark request dump somewhere.

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• VLC stream with trickplay

  - by marjasin

  The idea is to start a video stream from one computer and watch it on another with the ability to start/stop the stream. I think I could do this with VLC but i haven't been able to figure out how. I've tried the following: (From the official forum) Stream with RTSP and RTP: on the server, run: % vlc -vvv input_stream --sout '#rtp{dst=192.168.0.12,port=1234,sdp=rtsp://server.example.org:8080/test.sdp}' on the client(s), run: % vlc rtsp://server.example.org:8080/test.sdp But this doesn't give me the ability to start/stop the stream from the client. According to the VLC release note something called "Trick play" was added in version 1.0. This seems to be what I'm looking for but i can't find any documentation that descibes how to use it.

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• Video packet capture over multiple IP cameras

  - by nimals1986

  Hello We are working on a C language application which is simple RTSP/RTP client to record video from Axis a number of Cameras . We launch a pthread for each of the camera which establishes the RTP session and begins to record the packets captured suing the recvfrom() call... A single camera single pthread records fine for well over a day without issues.. but testing with more cameras available,about 25(so 25 pthreads), the recording to file goes fine for like 15 to 20 mins and then the recording just stops ..the application still keeps running .. Its been over a month and a half we have been trying with varied implementations but nothing seems to help .. Please provide suggestions.. We are using CentOS 5 platform

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• SDP media field format

  - by TacB0sS

  Hey, I would like to create a SDP media field with its attributes, and there are a few things I don't understand. I've skimmed and read the relevant RFC and I understand most of what each field means, but what I don't understand is how do I derive from the Audio/Video Format of the JMF, which parameters of the format compose the rtpmap registry entries I need to use. I see many times the fields m=audio 12548 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv these are received from the pbx server I'm connecting to, what do they mean in the terms of the JMF audio format properties. (I do understand these are standard audio format commonly used in telecommunication) UPDATE: I was more wondering about the format parameter '0 8 101' at the end of m=audio 12548 RTP/AVP 0 8 101 Thanks in advance, Adam Zehavi.

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• Configuring a PIX 506e for Asterisk

  - by orthogonal3

  Hi all! I'm having problems configuring a old Cisco PIX running 6.3 and wondered if anyone can lend a hand? Simply put I have a PIX 506e that I want to put in my VoIP data path. I can't update it and getting a compat version of Java for that version of PIX is tough so I can't log onto the web interface. The PIX straddles two networks..... 192.168.5.0 on the inside, ...50.0 on the outside both net masks are 255.255.255.0 I have a local Asterisk server cluster with a single service IP (<local asterisk>) SIP is on UDP 5060 and RTP (for the voip data) is on UDP 18000-18999 I know thats a big range but hey may as well. I need the 192.168.5.0 net to have web and ftp access for updates and the like. DHCP, DNS and NTP is already provided on that network so I don't need external DNS access. So I think I want the following rules: SIP or RTP from <my itsp> arriving at <outside voip ip> NATed to <local asterisk> SIP or RTP able to do the reverse route (should be covered by high sec - low sec??) HTTP and FTP access outbound for software update for the servers etc I have the following config at the minute - and I think I'm almost there (I hope)... interface ethernet0 auto interface ethernet1 auto nameif ethernet0 outside security0 nameif ethernet1 inside security100 enable password wouldyouliketobeapeppertoo encrypted passwd wouldyouliketobeapeppertoo encrypted hostname afirewall domain-name adomain fixup protocol dns maximum-length 512 fixup protocol ftp 21 fixup protocol h323 h225 1720 fixup protocol h323 ras 1718-1719 fixup protocol http 80 fixup protocol rsh 514 fixup protocol rtsp 554 fixup protocol sip 5060 fixup protocol sip udp 5060 fixup protocol skinny 2000 fixup protocol smtp 25 fixup protocol sqlnet 1521 fixup protocol tftp 69 access-list acl_ping permit icmp any any access-list voip permit ip host <my itsp> host <local asterisk> mtu outside 1500 mtu inside 1500 ip address outside <outside pix ip> 255.255.255.0 ip address inside <inside pix ip> 255.255.255.0 arp timeout 14400 global (outside) 1 <outside generic ip> nat (inside) 1 192.168.5.0 255.255.255.0 0 0 static (inside,outside) <outside voip ip> <local asterisk> netmask 255.255.255.255 0 0 static (outside,inside) <local asterisk> <outside voip ip> netmask 255.255.255.255 0 0 access-group acl_ping in interface outside access-group acl_ping in interface inside route outside 0.0.0.0 0.0.0.0 <my next hop router> 1 route outside <my itsp> 255.255.255.255 <my next hop router> 1 I think I just need a hand with the access-lists and NAT/static rules. Would anyone be able to help as I've RTFM'd the Cisco docs a few times and they're heavy. Wishing I'd completed my CCNA now! Thanks all for any help, Phil

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• QoS for Cisco Router to Prioritize Voice and Interactive Traffic

  - by TJ Huffington

  I have a Cisco 891W NATing Voice and Data to the internet over a 10mbit/2mbit connection. Voice traffic gets degraded when I upload large files. Pings time out as well. I tried to configure a QoS policy but it's basically not doing anything. Voice traffic still degrades when upload bandwidth gets saturated. Here is my current configruation: class-map match-any QoS-Transactional match protocol ssh match protocol xwindows class-map match-any QoS-Voice match protocol rtp audio class-map match-any QoS-Bulk match protocol secure-nntp match protocol smtp match protocol tftp match protocol ftp class-map match-any QoS-Management match protocol snmp match protocol dns match protocol secure-imap class-map match-any QoS-Inter-Video match protocol rtp video class-map match-any QoS-Voice-Control match access-group name Voice-Control policy-map QoS-Priority-Output class QoS-Voice priority percent 25 set dscp ef class QoS-Inter-Video bandwidth remaining percent 10 set dscp af41 class QoS-Transactional bandwidth remaining percent 25 random-detect dscp-based set dscp af21 class QoS-Bulk bandwidth remaining percent 5 random-detect dscp-based set dscp af11 class QoS-Management bandwidth remaining percent 1 set dscp cs2 class QoS-Voice-Control priority percent 5 set dscp ef class class-default fair-queue interface FastEthernet8 bandwidth 1024 bandwidth receive 20480 ip address dhcp ip nat outside ip virtual-reassembly duplex auto speed auto auto discovery qos crypto map mymap max-reserved-bandwidth 80 service-policy output QoS-Priority-Output crypto map mymap 10 ipsec-isakmp set peer 1.2.3.4 default set transform-set ESP-3DES-SHA match address 110 qos pre-classify ! fa8 is my connection to the internet. Voice traffic goes over a VPN ("mymap") to the SIP server. That's why I specified "qos pre-classify" which I believe is the way to classify traffic over the VPN. However even when I ping a public IP while saturating upload bandwidth, the latency is exceptionally high. Is this configuration correct? Are there any suggestions that might make this work for my setup? Thanks in advance.

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• QoS for Cisco Router to Prioritize Voice and Interactive Traffic

  - by TJ Huffington

  I have a Cisco 891W NATing Voice and Data to the internet over a 10mbit/2mbit connection. Voice traffic gets degraded when I upload large files. Pings time out as well. I tried to configure a QoS policy but it's basically not doing anything. Voice traffic still degrades when upload bandwidth gets saturated. Here is my current configruation: class-map match-any QoS-Transactional match protocol ssh match protocol xwindows class-map match-any QoS-Voice match protocol rtp audio class-map match-any QoS-Bulk match protocol secure-nntp match protocol smtp match protocol tftp match protocol ftp class-map match-any QoS-Management match protocol snmp match protocol dns match protocol secure-imap class-map match-any QoS-Inter-Video match protocol rtp video class-map match-any QoS-Voice-Control match access-group name Voice-Control policy-map QoS-Priority-Output class QoS-Voice priority percent 25 set dscp ef class QoS-Inter-Video bandwidth remaining percent 10 set dscp af41 class QoS-Transactional bandwidth remaining percent 25 random-detect dscp-based set dscp af21 class QoS-Bulk bandwidth remaining percent 5 random-detect dscp-based set dscp af11 class QoS-Management bandwidth remaining percent 1 set dscp cs2 class QoS-Voice-Control priority percent 5 set dscp ef class class-default fair-queue interface FastEthernet8 bandwidth 1024 bandwidth receive 20480 ip address dhcp ip nat outside ip virtual-reassembly duplex auto speed auto auto discovery qos crypto map mymap max-reserved-bandwidth 80 service-policy output QoS-Priority-Output crypto map mymap 10 ipsec-isakmp set peer 1.2.3.4 default set transform-set ESP-3DES-SHA match address 110 qos pre-classify ! fa8 is my connection to the internet. Voice traffic goes over a VPN ("mymap") to the SIP server. That's why I specified "qos pre-classify" which I believe is the way to classify traffic over the VPN. However even when I ping a public IP while saturating upload bandwidth, the latency is exceptionally high. Is this configuration correct? Are there any suggestions that might make this work for my setup? Thanks in advance.

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• Migrating complex SVN branch hierarchy to Mercurial

  - by Christian Hang

  Our team has been using SVN for managing an application of decent size and over time a rather complex hierarchy of branches and tags has built up, which is following the basic standard layout for SVN repositories, but is more nested: |-trunk |-branches | |-releases | | |-releaseA | | `-releaseB | `-features | |-featureX | `-featureY |-tags |-releaseA | |-beta | `-RTP `-releaseB |-beta `-RTP (The feature branches are obviously temporary branches but we have to take them into consideration as it won't be feasible to close all of them at once in the near future) For several reasons but primarily because merges have been becoming an increasing pain, we are considering to switch to Mercurial. The main problem we are currently facing is migrating the existing code base without losing our history. I've tried several migration tools (e.g., yasvn2hg, hg convert and svn2hg) with yasvn2hg being the most promising, but none of them seem to be able to deal with nested hierarchies but they all assume that branches and tags are organized in one flat directory respectively. The choice between named branches or clones as the conversion target of old SVN branches is not a limiting factor in this case, as either solution would be appreciated. We are currently experimenting with both options and how they would fit into our current processes but haven't decided on one yet. I'd obviously be interested in recommendations or experiences with similar setups concerning that issue as well. So, what is the best way to convert a nested SVN branch hierarchy like this to Mercurial? Converting one branch at a time into a separate repository would be quite annoying and I am not sure if that would be the right approach in the first place, depending on how the tools handle historic merges and need to be aware of all other branches?

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• Android getting XML values

  - by Nils

  Hello, I have the following XML code, which I got by a UPnP device and like to get the res value - the RTSP URL. In this case rtsp://10.42.0.103:554/live.sdp How can I do this? I heard that Android has some built-in support for reading XML. Is that true? <DIDL-Lite xmlns="urn:schemas-upnp-org:metadata-1-0/DIDL-Lite/" xmlns:dc="http://purl.org/dc/elements/1.1/" xmlns:upnp="urn:schemas-upnp-org:metadata-1-0/upnp/"> <item id="11" parentID="1" restricted="1"> <dc:title>Network Camera Stream 1</dc:title> <upnp:class>object.item.videoItem</upnp:class> <res protocolInfo="rtsp-rtp-udp:*:video/mpeg4-generic:*" resolution="640x480">rtsp://10.42.0.103:554/live.sdp</res> </item> <item id="12" parentID="1" restricted="1"> <dc:title>Network Camera Stream 2</dc:title> <upnp:class>object.item.videoItem</upnp:class> <res protocolInfo="rtsp-rtp-udp:*:video/mpeg4-generic:*" resolution="176x144">rtsp://10.42.0.103:554/live2.sdp</res> </item> </DIDL-Lite>

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• Windows XP "automatic" services not starting

  - by Mala

  Hi I have a fresh install of WinXP. The main problem is that every time I start it up, I have to go into Administrative Tools and start the needed services, such as DCOM, RTP, DHCP, etc etc. The only services that start automatically are: plug and play remote procedure server windows audio workstation All of the rest have to be started manually, in spite of the fact that they're listed as "automatic". Why won't they start on their own like they should? Thanks, Mala

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