RTP and RTCP in JMF

I need to test VoIP application in a wireless
network. I have a program that reads a sound file and
creates a RTP session to send it to other computer.
In order to measure the application performace I need
to take the following metrics during the RTP
session:
- Packet loses.
- End to end delay
- Jitter
The questions are: how does JMF implementation of
RTCP can help with all this metrics?? Does someone
know some example code?
javax.media.rtp
Interface GlobalTransmissionStats
javax.media.rtp
Interface ReceptionStats
javax.media.rtp
Interface TransmissionStatsGo through the API to see which interfaces and classes can aid your needs.

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About rtp and jmf

Hello,I'm now working on my graduation paper which refers to the RTP,and JMF matter.Because this is my first time to deal with this, i don't know much about that.And what i'm now trying to transfer is a file like *.au by using the sample
code VideoTransmit.java. But it always says
-cannot create processor
(processor is an interface)
I don't know why.
can anyone help me?
Thank you.

.au is a sound file, VideoTransmitter wants a video file

Transmit RTP without RTCP!

Hi,
Ive written an application to Transmit and Receive RTP over Custom Transport Layer with the JMF 2.1, but I'm not able to transmit RTP without RTCP. Is there any idea how this could be done?
Any help will be appreciated!
Thanks in advance,
ARIF

Looks like you implement your own version of the RTPConnector interface, and then use that to handle the input/ouput streams manually rather than relying on the default RTPManager with the local and remote SessionAddress objects.

RTP and RTSP

I have 2 question want to ask:
1)Is RTP and RTSP can pass through firewall?
Or can you tell me which TCP port no. be used for RTP and RTSP protocol?
2)Also,Is this possible to transfer video data thro' http protocol?

1a: Yes , (If firewall rules permit)port 554 for RTSP (http://www.chebucto.ns.ca/~rakerman/port-table.html)
1b: No, (if firewall rules disallow)
2:(see answers above)

Does Cisco Media Services Interface mark RTP and Signaling packets regardless of Cisco Prime Collaboration/MediaNet Deployment?

Hi there,
I'm deploying Cisco UC Integration Lync 9.2(5). There's a QoS portion which states that you can install Cisco Media Services Interface on the user's PC and is to be used with Cisco Prime Collaboration or MediaNet:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucilync/9/CUCI_BK_C0545A41_00_cisco-uc-integration-for-microsoft/CUCI_BK_C0545A41_00_cisco-uc-integration-for-microsoft_chapter_010.html#CUCI_CN_IDBE72E9_00
If we don't have Cisco Prime Collaboration or MediaNet, does Cisco Media Services Interface mark RTP and Siganling packets on the outbound regardless? Or do I have to manually set the DSCP values for the PC?
Thanks,
John

When you were placing the internal test calls, where was your phone located? The RTP streams are usually only set up between the two endpoints (assuming no transcoder or MTP is used).
Have you have already checked calls between IP phones at your central site (where Unity is located) and IP Phones at site A? Are these working fine? If not, then it's likely you do have a QoS issue over the WAN to Site A.
Have you confirmed whether or not calls from the PSTN gateway at site A experience any voice QoS issues when transferred to an IP Phone at the central site?
Is the same codec is being used for the MGCP gateway as for IP Phones at that site (check the Region settings for the Device Pool it's in)?
If these things have all been checked, then you may want to consider collecting a simultaneous sniffer trace from each end of a single call, and comparing what's seen at the two endpoints.
Kind regards,
Craig

Packet sniffer only picks up UDP and no RTP packets when using JMF???

Hi,
I am developing a voice mail application to interface with asterisk. Here is the problem.
I am using ethereal packet sniffer to sniff the packets. When I connect two regular SIP phones and sniff , I can sniff the RTP packets.
But when I use JMF AVtransmit2.java and AVReceive2.java I sniff only UDP packets and no RTP packets.
I am very confused. What is going on? If JMF sends over RTP (that uses UDP underneath), then why cannot packet sniffers detect it.

Hi,
I am developing a voice mail application to interface with asterisk. Here is the problem.
I am using ethereal packet sniffer to sniff the packets. When I connect two regular SIP phones and sniff , I can sniff the RTP packets.
But when I use JMF AVtransmit2.java and AVReceive2.java I sniff only UDP packets and no RTP packets.
I am very confused. What is going on? If JMF sends over RTP (that uses UDP underneath), then why cannot packet sniffers detect it.

JMF , RTP and IPV6

Hello! How are you? I need to know if it is possible with the functions of JMF of RTP to transmit video and sound but using a direction IPV6� and if there is to modify something by where to begin� Thank you very much! So long.

More debugging, but still at a block:
It seems: com.sun.media.rtp.RTPSessionMgr is instantiated as the rtpmanager.
There is no source for this and so cannot figure out why this fails.
Best regards,
-Arun.

JMF, RTP and SIP

Hi,
I am building an application that shall send streaming data to a mobile client (personaljava) with jmf. I read that RTP is blocked by many firewalls, so I wonder if it is possible to use an alternative protocol (SIP?) with advantages compared to RTP, still using JMF.
Any hint appreciated..

SIP is a signaling protocol and as far as I know it does not transport the media, it uses rtp to do so.

JMF RTP and Linux/Fedora Core3

Hello,
I am attempting to use JMF 2.1.1e with Linux/Fedora Core3, and see an exception as:
java.io.IOException: Can't open local data port: 51450
at com.sun.media.datasink.rtp.Handler.open(Handler.java:141)
Debugging more, I can see the Handler.open fails when it is trying to
rtpmanager.initialize(localaddr);
where localaddr is "SessionAddress()" (no args to the constructor).
I am unable to debug beyond this. Any help?
Best regards,
-Arun.

More debugging, but still at a block:
It seems: com.sun.media.rtp.RTPSessionMgr is instantiated as the rtpmanager.
There is no source for this and so cannot figure out why this fails.
Best regards,
-Arun.

JMF stream over RTP and VLC

Hi All!
I wrote an application that streams video (without audio) using h263 codec over RTP.
Does anybody know how to play this stream in VLC?
Second issue I am struggling with is to make SDP file that will describe this stream. Any help?
Thank you.
Edited by: 814312 on 22.11.2010 2:05

814312 wrote:
Does anybody know how to play this stream in VLC?Sounds like a VLC question, not a JMF one...
Second issue I am struggling with is to make SDP file that will describe this stream. Any help? Never heard of a SDP file, so I'd assume this isn't really a JMF question either ;-)

RTP and sessions please help me

I been reading de guide of JMF, but I can�t understand something about RTP.
I want to create videoconference, and my question is, I have to transmit to destination IP? and what have to do the client.
I very disoriented, can someone explain me or send some complete example for understand the process.
thanks ( sorry my english)

AVTransmit2.java will transmit to specific IP:port (port must be even)
It will open same port on server as that you transmit to.
RTCP port will automatically be +1 of the port;
AVReceive2.java will receive on specific port.
I learned by running those examples on JMF solutions page, and then use Network monitoring tool(ETHEREAL) to understand the ports and data that is transmitted.
http://java.sun.com/products/java-media/jmf/2.1.1/solutions/AVReceive.html
http://java.sun.com/products/java-media/jmf/2.1.1/solutions/AVTransmit.html

Vaibility of RTP relay server using JMF

Hello!
I am thinking of making an RTP server that would just relay the streams received from single user to multiple users.
I want to ask following questions:
1- Is it viable to make such a Server using JMF?
2- If it is, then how should I proceed?
3- Should I need to create Processor for each recieved stream at the server or just cloning the DataSource received and sending them using separate RTPManagers would solve my problem?
4- Is cloning of data source needed at all?
I am asking this before doing any coding just in case if it is not possible than you warn me ,and because I had real bad experiences while cloning the data source and this server, I think depends on cloning.
Else, I want some help regarding code. I would highly appreciate some code snippets.
Thanks in advance.
Thanks!
P.S.: captfoss, Are you listening?

Now some simple questions from a novice, thats me:I will answer them out of order
3- Are these terms specific to JMF or are they general Networking terms?They are general networking terms.
2- What is difference b/w unicasting and multicasting?Uni = Latin prefix for "one".
Multi = Latin prefix for "many"
Broad = Latin prefix for "all" (okay, this one is probably not true, but...)
unicast = sending data to a single recipient.
broadcast = sending data to all recipients.
multicasting = sending data to many recipients.
It deals with how the underlaying UDP packets are handled.
Unicast addresses the packets to a specific host, so all other hosts that receive those packets go "Not for me" and ignore it.
Broadcast addresses the packets to a special IP address, the broadcast ip, so all hosts that receive it say "Oh, this is a broadcast message so it's for me"
Multicast addresses the packets to an IP address in a special range (Class D addresses), and all hosts can opt to join in to the "multicast session". If they join the multicast session, it basiclly means when they receive packets addressed to any multicast addresses that they have joined the session of, they will consider those packets to be "for them".
1- What exactly is multicasting in JMF?JMF multicasting is basiclly where the "host" can send out a single stream, and any number of "clients" can receive the stream.
4- How multicasting is handled at Transmitter and Reciever side using JMF, some java statements please.Multicasting is almost handled "automaticlly".
It's handled by giving the transmitter a multicast IP address to send to. "224.123.123.123" is an example of the one I always used for testing (because it was easy to remember). Transmitting multicast packets is handled automaticlly.
Receiving multicast packets requires a little special handling.
From AVReceive2.java
if( ipAddr.isMulticastAddress()) {
    // local and remote address pairs are identical:
    localAddr= new SessionAddress( ipAddr,
                       session.port,
                       session.ttl);
    destAddr = new SessionAddress( ipAddr,
                       session.port,
                       session.ttl);
} else {
    localAddr= new SessionAddress( InetAddress.getLocalHost(),
                            session.port);
    destAddr = new SessionAddress( ipAddr, session.port);
}The main difference here is that your "local" address isn't going to be LocalHost address, "127.0.0.1", it's going to be the multicast address.
And you should define the TTL (time to live) for the SessionAddress because multicast packets can only travel a certain number of "hops" (number of times forwarded).
But honestly, I'm pretty sure I ripped that IF statement out and multicasting still worked (but I define the TTL for unicast as well, so bear that in mind...)
Ignoring all of the stuff above, multicasting is a great idea for a local LAN application where there are no routers between the host and the clients. If you've got a router between them, then the router may not forward the multicast packets and your stream may never get to the remote clients.
That's why in my project, which is web based, clients attempt to get the multicast packets and if they fail at that, then they request a unicast from my server.

Cretae a RTP packet without using JMF

I have to create a RTP packet of existing audio/video file using all the parameters of RTP Packet
I dont have to use JMF.
Please suggest me the soluion and if possible provide me the codes.
Thanks alot.

I have to create a RTP packet of existing audio/video file using all the parameters of RTP Packet
I dont have to use JMF.
Please suggest me the soluion and if possible provide me the codes.
Thanks alot.

RTP and applets

Since rtp is a push protocol (unicast, multicast, broadcast)...what to do when you are connecting via an applet from a dynamic ip?
Does one need to 'register' one's ip (of the applet) and restart the video server (i.e. jmf)?
Since server requires destination address this must be pre-known, but when you are connecting from some coffeehouse browser...and maybe via dhcp with a changing ip, how does this happen?
I've got applets on the same network showing rtp video...however for dynamic ips outside of this network???

you should get the IP of the visitor through the Webserver. There are different methods in different languages like ASP/PHP to get this done ... once you get the IP of your visitor you can send the stream to that IP. I m gona give an example of ASP for your help.
Request.ServerVariables("REMOTE_HOST")this returns the IP
have fun .. since ur using Applets i might add my question here, JMF is very large to download if do not already have it installed on ur system when someone visites your webpage (with applet using JMF), is there anyway to customize the size of this JMF.
NOTE that i need to include the sound capturing feature in my applet too, so cant use the CROSSPLATFORM version of JMF :).
THANKS

How to hide RTPs and Specify Fix value for them

Hi All,
I created a business in which i am using one variable of String type with RTP enabled.
Now i don't want my user to enter this value while executing business rule, instead i want to provide a fix/constant value for that variable. How can i do that.
Basically my intention is to put that business rule inside a rule twice with 2 diff values

Have you tried setting the variable as a member type, applying it against a dimension and then giving it a default value.
Cheers
John
http://john-goodwin.blogspot.com/

RTP and RTCP in JMF

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