Lead Addresses in IS-Media

I need to validate postal addresses entered into SAP IS-Media (industry solution for media). According to the product manager:
...IS-M has its own address management which is independant from the one used in the SAP business partner. We are updating the SAP business partner adress (depending on the role), but we are not using the 'standard' (core) address managment. The leading address is the one used in IS-M...
We have integrated with ADDRESS_CHECK BAdI (Business Add-In). If it's not using ADDRESS_CHECK BAdI, how can I validate the "leading address... used in IS-M"?
Thanks in advance.

Hi, can u please send me some material related to IS-media.. coz i am new to the module and i'm not aware of the basics of this module...
thanks in advance

Similar Messages

Help! How do i get rid of self-assigned IP address? Virgin Media & Ibook

Hey all, first time here. Hoping someone more knowledgeable than me (or the people at virgin!) can help.
I have home broadband from Virgin as of last night. Self-set up the modem (Cisco), Router (D-link) and connection to my Ibook, running Tiger 10.4.11
Router and modem seem to be working fine. Green lights all round. Plus last night i had an ethernet connection runnign fine and even got a wireless connection to the Ibook. All good.
This morning, twenty minutes of internet use, then bumped off. No connection.
Checked under System Preferences, Network, Airport and got a message saying Airport has 'a self assigned IP address and cannot connect to internet.'
I assume the problem therefore is with the Ibook talking to the router (sorry, not very technical so 'talking' is best i can do).
Tried switching off the firewall. Still no joy. Airport can see my router in the list, and even has it first in preferred networks to join list. I get a five bar signal when i select it from Airport but how do i rid the computer of the 'Self-assigned IP address' ?
Please help, it's driving me mad! Apple blame Virgin, Virgin blame Router manufacturer, Router manufacturer blame Apple. It's an endless circle of **** and call-centres!!!
Cheers!
Dan

Hi Dan, and a warm welcome to the forums!
"self-assigned IP address" simply means it's not getting an IP from the device responsible for handing out IPs, the Router in your instance.
Does Ethernet still work fine?
Go to System Preferences
Click Network
Highlight AirPort and click Configure...
Choose “By default, join: Preferred networks”
Select your access point and Remove your access point with the minus button.
Launch your keychain access in Utilities and delete your access point keychain entry.
Reboot
Go back to the “By default, join:” page and click the plus this time to add your access point. Enter the correct password, save, reboot.

How can I view the IP or DNS address for Adobe Media Server

Hello, I am trying to use AMS to conduct a live school news broadcast. Currently I can view the stream using `localhost,` but how can I figure out what IP or DNS address to use to make the stream public. While I can stream on the same network the viewers are on, I'd like to make it visible to anyone.

Are you streaming from your own PC?!
For local tests, you can get your ip address by opening a command line and running ipconfig.
Assuming that your PC's firewall doesn't block the relevant ports, then others on your network could see it by replacing "localhost" with that ip.
For release... you may want to move your server.
G

HT204053 Can I create another email address with icloud. Like we are able to create more then 5 email address in virgin media. Help

Can I create another email address with icloud.com. Like we can create few email on virginmedia or other provider

You can add up to three 'email aliases' - these are additional addresses (not accounts) which deliver into the same inbox as the main account. (In fact it's a good idea to give out alias addresses, rather than the main address, because if they attract spam you can easily change them.) (New aliases can only be @icloud.com ones; @me.com addresses cannot now be created.)
You should be aware before you start that once you've created an alias you cannot turn that address into a full iCloud account or move it to another account.
More information on aliases here: http://help.apple.com/icloud/#mm6b1a490a
You can of course create additional iCloud accounts, but any one device can create only three accounts - after that it can sign into accounts created on another device but not create any more (this is tied to the hardware).

QT Broadcasting through media server question

Hey all,
My question is this: does anyone know how (if its even possible) to use QT Broadcaster on MANUAL UNICAST to webcast to users connecting to a media server?
for instance: G5 ---> media server <----user
<----user
<----user etc.
I have been trying to set up the webcast as such: creating a manual unicast .sdp with the IP address of the media server contained within. upload the .sdp to the media server where the users will find it and connect to the stream.
So far, all I get on the client side is a QT window that says, "Live Broadcast" and "ready".
If the clients connect to the stream through the media server (which has its own static IP that matches the one in the .sdp) shouldn't they be able to view the webcast?
Any and all thoughts are most appreciated.
-=adam=-
[email protected]

Using Manual Unicast:
Manual unicast is a method of broadcasting that requires you to export an SDP file from QuickTime Broadcaster to one receiving computer identified by its IP address. The receiving application can be a single QuickTime Player or QuickTime Streaming Server or Darwin Streaming Server. If the receiving application is QTSS or DSS, the broadcast is reflected and no longer limited to a single IP address.
To broadcast via Manual Unicast:
If necessary, click Show Details in the QuickTime Broadcaster window, and in the Network pane choose a preset.
Choose Manual Unicast from the Transmission pop-up menu, and enter the IP address of the receiving computer (or accept the default IP address of your own computer if broadcasting to yourself as a test), the audio port and video port numbers (or accept the defaults if broadcasting one-to-one), and the buffer delay (or accept the default).
If desired, also enter information in the Annotation section.
Click Broadcast, and then choose Export from the File menu and SDP or Movie from the Export submenu.
Specify a name for the SDP or movie file and click Export.
If the file is exported directly to a QuickTime Player, the viewer opens the file using "Open Movie in New Player" to view the broadcast. If the file is exported to QTSS or DSS, the viewer instead uses "Open URL in New Player." Talk to your QTSS or DSS administrator to determine the correct URL.
Viewing a QT broadcast:
Your audience can view your live broadcast using either QuickTime Player or a web browser with the QuickTime Plugin. You determine how users will see the broadcast by distributing an announcement movie file, which is needed by QuickTime Player to "tune in" to your broadcast, or by creating a Web page that refers to the movie file.
For best results, your viewers should use QuickTime 6. QuickTime Broadcaster streams can be viewed using versions as early as QuickTime 4, but some viewers may not be able to watch your broadcast if you choose compression methods that are not available in the older software.
When viewers connect to your stream, progress messages appear about connecting, negotiating, and buffering, then the stream begins playing. Viewers can pause or stop the broadcast, but when they resume they'll jump to the current moment if they're watching a live broadcast.
If viewers are watching an archived (or prerecorded) broadcast, they will be able to pause, rewind, or skip ahead using the QuickTime controls.
Annotating your broadcast:
You can add a title, copyright, and other annotations to your broadcast.
If necessary, click the Show Details button in the QuickTime Broadcaster window. Then click the Network tab and fill in the desired fields under Annotations.
Your audience can view this information by choosing Show Movie Info from the File menu in the QuickTime Player application.
About Instant-On streaming:
Viewers watching a streaming video with QuickTime 6 will benefit from Instant-On, an advance in Apple's patent-pending Skip Protection technology that dramatically reduces buffer, or wait, time for an instantaneous viewing experience. Viewers can click around or scrub video as if it was on their hard disk.
Users must have a broadband connection to experience Instant-On. The responsiveness of Instant-On is affected by available bandwidth and size, or data rate, of the content. It can also be affected by the codec used. Hinted movies made using the QDesign Audio codec, for example, may not be Instant-On capable.
Choosing a broadcast method:
The size of your expected audience determines how you deliver the broadcast.
For example, when broadcasting a live event to only one QuickTime Player, QuickTime Broadcaster can send a single stream using Mac OS X without server software.

How do I configure AirportExtreme or TimeCapsule to use UPnP or connect an iOmega Media Drive, Could Edition?

I recently purchased an iOmega Home Media Drvie, Cloud Edition, which claims functionality with the MacOS--and shared foldersdo appear on desktops connected the home network (wirelessly or otherwise)—however this puprose of this "Cloud Edition" is to be able to share files/the drive over the internet and I cannot seem to configure the Apple AirPort TimeCapsule to allow this (other routhers are configured automatically by the drive).
The documentation says "you’ll need to turn on Remote Access and enable UPnP on your router" and
The Home Media Drive tests for an active internet connection. If this test fails, check your network configuration.
The Home Media Drive will identify your WAN IP address. The WAN IP address is the extrnal address of your router or internet gateway and is the address your Home Media Drive will be identified by on the internet. If this test fails you will need to double check the network configuration of your Home Media Drive and your Router.
The Home Media Drive will test to make sure UPnP port forwarding is operating correctly. If this test fails, you will need to manually port forward your router.
Once the testing is complete, The device will be available on the internet via its WAN IP address, i.e. https://168.254.13.68. See the WAN IP Field for the exact address.
I can't figure out out how to do this, and I am not a networking dummy. Any help would be appreciated!
Thank you!

If you have the Airport Extreme or TimeCapsule connected to another router from your ISP it can cause a double NAT situation. You'd have to first port forward the primary router (from your ISP) to the secondary router using the same 50500 port that you are using to port forward the AirPort Extreme to the Iomega drive. You want to be sure that you only have one router with DHCP and NAT enabled. So the simplest solution is to turn the AirPort Extreme or TimeCapsule to a bridge. Then restart all devices. I reset the iOmega media drive after switching to bridge mode on the AirPort Extreme and the cloud was enabled and finally configured properly. You can reset without loosing your data... but you will have to reconfigure all users including administrators. Same with remote access. Hope this helps.

Web page music loads in IE but not Firefox. Have downloaded Windows Media Player Firefox plug in but still nothing. Any other suggestions?

Can't get any sound when I open one webpage in Firefox but sound works in Internet Explorer. sound files are .wmv and mp3 format.
I have Windows Media Player Firefox Plug in installed and set to always activiate. Also have Quicktime & Shockwave activated. Any suggestions. Operating system is Windows 7. Thanks

There are a couple issues here that might be the culprit. One is that the player is hidden using a variety of techniques and Firefox often suppresses hidden content because it can be annoying not to be able to control your experience with media. Another possible issue is that the embed tag doesn't specify a content type. Firefox appears to be guessing anyway, and in my case asking whether to activate the QuickTime plugin, but then nothing plays.
To test what would happen if both of those were addressed, you can run this script using Firefox's Web Console. It grabs the address of the media and creates a new embed that is visible and specifies the content type. Here's how to use it:
Select the following long line of code and copy it:
function reEmbedMpeg(em){var el=document.createElement("embed"); el.setAttribute("type","audio/mpeg"); el.setAttribute("src",em.src); el.setAttribute("style","height:20px;width:200px;position:fixed;right:0;top:0"); document.body.appendChild(el);} reEmbedMpeg(document.querySelector("embed"));
When viewing the problem page, press Ctrl+Shift+k to open the Web Console in the lower part of the tab.
Paste the code next to the caret (>>) on the bottom line and press Enter to run it. A player should appear in the upper left corner, the progress bar should show downloading and then the content should play.
Obviously it's not practical to hack every web page you find, but if this is your site or you would be interesting in contacting the owner, hopefully this post will help in getting this working in Firefox.

SRW248G4P and media center extender

I have a SRW248G4P 48 port POE switch and a few DMA2100 Media Center Extenders all hardwired. When everything is connected to the switch the extenders cannot connect or be detected by the Windows 7 Media Center PC. However, when I switch all of the cabling over to the 4 ports on the router (RV042) everything connects and operates as usual.
Any Ideas as to what the problem my be?
Thanks,
Aaron

Did you assign static IP address to your Media extender? If not that may be the reason why it is not working. This is because, RV042 has DHCP capability that assign IP address to its client. On the other hand SRW248G4P doesn't have DHCP capability and if you connected your DMA2100 devices to the switch it will not get any IP address to be part of the network.

Destination for media files changed unexpectedly

My iTunes media files are stored on a NAS. Recently, for the second time, I have seen (annoyingly after importing new music files) that the address specified for the iTunes Media folder location in iTunes Preferences has somehow changed to point to the default value on the internal disk. The first time this happened I thought that it might have been due to the NAS not being accessible when iTunes was started (though I don't find it acceptable that the address is changed without any warning). As far as I know, this wasn't the case when the problem arose for a second time. Both times the problem was on an iMac running Mavericks. I haven't had the trouble (as far as I know) on my older iMac with Snow Leopard.
Has anyone else had similar trouble or an explanation for the problem?

As soon as I had posted this I found this other thread that probably explains the problem: https://discussions.apple.com/message/18703603#18703603
Any other tips gratefully accepted, though. I still don't find it right that the address of the media folder is changed without any warning.

Can flash media live encoder feed stream to remote adobe media server installed on other machine?

Hi All,
i have installed Flash Media Live Encoder 3.2 on my one machine xxx and AMS version 5 on another machine yyy.
i am trying to configure FMLE for ams server URL - i.e. stream to flash media server option on FMLE. I have generated FMS URL as "rtmp://yyy/live" and clicked "connect" to. Its not getting connected to it.
please find attachment for error.
is there any restriction that our AMS should be on same machine wehere we have installed flash media live encoder ??
please help on this!!

If you have any trouble to understand from above query then please go through following scenarios and do reply if you have answers to it.
I have installed Flash Media Live Encoder 3.2 on my machine and Adobe Media Server 5.0.3 on another machine.
My requirement is to stream RTMP from Flash Media Live Encoder which is installed on my machine(source) and receive RTMP stream on Adobe Media Server which is installed on another machine.
I am unable to connect to "rtmp://<"IP Address of Adobe Media Server installed system">/live on AMS installed machine.
Atached please find the error message screen shot:
Are there any restriction that Adobe Media Server and Flash Media Live Encoder must be installed on the same machine?

[Mac OS X] How to detect media status of a network interface?

Hi
I'm looking for a way to see the status of a network interface, both wire and wireless, in Mac OS X. The older OS X version to support it the better.
I'm not any experienced Mac developer so forgive me for not knowing my way around in development for this platform.
In plain English I want to detect in programming (language is C) if the cable is plugged in or not. In wireless environment it means if the computer is associated to a wireless network.
It should not depend on if the computer has an IP-address, just the media status.
It would be good to see how many interfaces that are connected but just getting the answer to question if there is any (real, not loopback) interface connected would be good enough.
When searching on internet I found people using 'grep' on the output from 'ifconfig'. As I'm not doing a shell script I don't like a solution where I would call 'ifconfig' and take the output back into my program.
The most promising part I've found is in Core Foundation, the CFNetDiagnostics. But it seems to operate on a higher level (streams?), which is not what I want.
If any of you are familiar with windows development I found the IsNetworkAlive function in SENS API, http://msdn.microsoft.com/en-us/library/aa377522.aspx
It is something similar to that is what I'm looking for.
The program in question is a Java application but Java is very bad at low level network programming so I have to use native interface to windows and OS X.
Thanks in advance
Christer

Update:
I found SCNetworkConfiguration.h
http://developer.apple.com/documentation/Networking/Reference/SysConfig/SCNetwor kConfiguration/index.html
Seems to fit the description, do anyone agree?
Basically I have to get all the interfaces, then for each one list its active media options and see what I find.

WMA Player Plug-in in Floating DIV

Here is a 'test' page that I've created:
http://www.justus.ca/test/vocab/professions.html
It has a 'floating' Windows Media Play plug-in within a
floating div at the
upper left of the screen. I can not get it to play, so I
copied the player
and pasted it into the first cell of the table as a 'test'.
That player
plays fine. It is an exact duplicate of the player in the
floating div
other than where it is located. Can anyone tell me how I can
get the
floating player to work? I want the player to stay fixed on
the screen
while people scroll the page to follow the audio being
played.

Try using this website
http://cit.ucsf.edu/embedmedia/
Note its easily to test an online media file then a local
media file I find.
For online files you can simply type the web address into the
media player script
for local file it has to be something like this src=file://
etc..
not just src=file.mp3
so try to upload you music or video file onto the internet
then
use the generator on the website
easy peasy
Bookmark it its the best around....
Daniel Lee
| CEO - Lead Developer
| Developer Trainer
|
http://www.onemegamarket.com

Problem with using JMF audio over a network

Hiya, I'm using IBM JMF code but I'm having problems trying to get it transmit data from the MediaTransmitter to the MediaPlayerFrame.
I'm kinda new to JMF so I assume I'm missing something basis for why this doesn't work.
Any help would be greatly appreciated.
MediaPlayerFrame
import javax.media.*;
import javax.swing.*;
import java.awt.*;
import java.awt.event.*;
import java.io.*;
* An instance of the MediaPlayerFrame may be used to display any media
* recognized * by JMF. This is intended to be a very simple GUI example,
* displaying all possible controls for the given media type.
public class MediaPlayerFrame extends JFrame {
     * The frame title.
    private static final String FRAME_TITLE = "developerWorks JMF Tutorial " +
        "Media Player";
     * The panel title of the main control panel.
    private static final String CONTROL_PANEL_TITLE = "Control Panel";
    // location and size variables for the frame.
    private static final int LOC_X = 100;
    private static final int LOC_Y = 100;
    private static final int HEIGHT = 500;
    private static final int WIDTH = 500;
     private final static long serialVersionUID = 1;
     * The current player.
    private Player player = null;
     * The tabbed pane for displaying controls.
    private JTabbedPane tabPane = null;
     * Create an instance of the media frame. No data will be displayed in the
     * frame until a player is set.
    public MediaPlayerFrame() {
        super(FRAME_TITLE);
        System.out.println("MediaPlayerFrame");
        setLocation(LOC_X, LOC_Y);
        setSize(WIDTH, HEIGHT);
        tabPane = new JTabbedPane();
        getContentPane().add(tabPane);
        /* adds a window listener so that the player may be cleaned up before
           the frame actually closes.
        addWindowListener(new WindowAdapter() {
                               * Invoked when the frame is being closed.
                              public void windowClosing(WindowEvent e) {
                                  closeCurrentPlayer();
                                  /* Closing this frame will cause the entire
                                     application to exit. When running this
                                     example as its own application, this is
                                     fine - but in general, a closing frame
                                     would not close the entire application.
                                     If this behavior is not desired, comment
                                     out the following line:
                                  System.exit(0);
     * Creates the main panel. This panel will contain the following if they
     * exist:
     * - The visual component: where any visual data is displayed, i.e. a
     * movie uses this control to display the video.
     * - The gain component:   where the gain/volume may be changed. This
     * is often * contained in the control panel component (below.)
     * - The control panel component: time and some extra info regarding
     * the media.
    private JPanel createMainPanel() {
        System.out.println("createMainPanel");
        JPanel mainPanel = new JPanel();
        GridBagLayout gbl = new GridBagLayout();
        GridBagConstraints gbc = new GridBagConstraints();
        mainPanel.setLayout(gbl);
        boolean visualComponentExists = false;
        // if the visual component exists, add it to the newly created panel.
        if (player.getVisualComponent() != null) {
            visualComponentExists = true;
            gbc.gridx = 0;
            gbc.gridy = 0;
            gbc.weightx = 1;
            gbc.weighty = 1;
            gbc.fill = GridBagConstraints.BOTH;
            mainPanel.add(player.getVisualComponent(), gbc);
        // if the gain control component exists, add it to the new panel.
        if ((player.getGainControl() != null) &&
            (player.getGainControl().getControlComponent() != null)) {
            gbc.gridx = 1;
            gbc.gridy = 0;
            gbc.weightx = 0;
            gbc.weighty = 1;
            gbc.gridheight = 2;
            gbc.fill = GridBagConstraints.VERTICAL;
            mainPanel.add(player.getGainControl().getControlComponent(), gbc);
        // Add the control panel component if it exists (it should exists in
        // all cases.)
        if (player.getControlPanelComponent() != null) {
            gbc.gridx = 0;
            gbc.gridy = 1;
            gbc.weightx = 1;
            gbc.gridheight = 1;
            if (visualComponentExists) {
                gbc.fill = GridBagConstraints.HORIZONTAL;
                gbc.weighty = 0;
            } else {
                gbc.fill = GridBagConstraints.BOTH;
                gbc.weighty = 1;
            mainPanel.add(player.getControlPanelComponent(), gbc);
        return mainPanel;
     * Sets the media locator. Setting this to a new value effectively
     * discards any Player which may have already existed.
     * @param locator the new MediaLocator object.
     * @throws IOException indicates an IO error in opening the media.
     * @throws NoPlayerException indicates no player was found for the
     * media type.
     * @throws CannotRealizeException indicates an error in realizing the
     * media file or stream.
    public void setMediaLocator(MediaLocator locator) throws IOException,
        NoPlayerException, CannotRealizeException {
          System.out.println("setMediaLocator: " +locator);
        // create a new player with the new locator. This will effectively
        // stop and discard any current player.
        setPlayer(Manager.createRealizedPlayer(locator));
     * Sets the player reference. Setting this to a new value will discard
     * any Player which already exists. It will also refresh the contents
     * of the pane with the components for the new player. A null value will
     * stop the discard the current player and clear the contents of the
     * frame.
    public void setPlayer(Player newPlayer) {
        System.out.println("setPlayer");
        // close the current player
        closeCurrentPlayer();
        player = newPlayer;
        // refresh the tabbed pane.
        tabPane.removeAll();
        if (player == null) return;
        // add the new main panel
        tabPane.add(CONTROL_PANEL_TITLE, createMainPanel());
        // add any other controls which may exist in the player. These
        // controls should already contain a name which is used in the
        // tabbed pane.
        Control[] controls = player.getControls();
        for (int i = 0; i < controls.length; i++) {
            if (controls.getControlComponent() != null) {
tabPane.add(controls[i].getControlComponent());
* Stops and closes the current player if one exists.
private void closeCurrentPlayer() {
if (player != null) {
player.stop();
player.close();
* Prints a usage message to System.out for how to use this class
* through the command line.
public static void printUsage() {
System.out.println("Usage: java MediaPlayerFrame mediaLocator");
* Allows the user to run the class through the command line.
* Only one argument is allowed, which is the media locator.
public static void main(String[] args) {
try {
if (args.length == 1) {
MediaPlayerFrame mpf = new MediaPlayerFrame();
/* The following line creates a media locator using the String
passed in through the command line. This version should
be used when receiving media streamed over a network.
mpf.setMediaLocator(new MediaLocator(args[0]));
/* the following line may be used to create and set the media
locator from a simple file name. This works fine when
playing local media. To play media streamed over a
network, you should use the previous setMediaLocator()
line and comment this one out.
//mpf.setMediaLocator(new MediaLocator(
// new File(args[0]).toURL()));
mpf.setVisible(true);
} else {
printUsage();
} catch (Throwable t) {
t.printStackTrace();
MediaTransmitter
import javax.media.*;
import javax.media.control.*;
import javax.media.protocol.*;
import javax.media.format.*;
import java.io.IOException;
import java.io.File;
* Creates a new media transmitter. The media transmitter may be used to
* transmit a data source over a network.
public class MediaTransmitter {
     * Output locator - this is the broadcast address for the media.
    private MediaLocator mediaLocator = null;
     * The data sink object used to broadcast the results from the processor
     * to the network.
    private DataSink dataSink = null;
     * The processor used to read the media from a local file, and produce an
     * output stream which will be handed to the data sink object for
     * broadcast.
    private Processor mediaProcessor = null;
     * The track formats used for all data sources in this transmitter. It is
     * assumed that this transmitter will always be associated with the same
     * RTP stream format, so this is made static.
    private static final Format[] FORMATS = new Format[] {
        new AudioFormat(AudioFormat.MPEG_RTP)};
     * The content descriptor for this transmitter. It is assumed that this
     * transmitter always handles the same type of RTP content, so this is
     * made static.
    private static final ContentDescriptor CONTENT_DESCRIPTOR =
        new ContentDescriptor(ContentDescriptor.RAW_RTP);
     * Creates a new transmitter with the given outbound locator.
    public MediaTransmitter(MediaLocator locator) {
        mediaLocator = locator;
     * Starts transmitting the media.
    public void startTransmitting() throws IOException {
        // start the processor
        mediaProcessor.start();
        // open and start the data sink
        dataSink.open();
        dataSink.start();
     * Stops transmitting the media.
    public void stopTransmitting() throws IOException {
        // stop and close the data sink
        dataSink.stop();
        dataSink.close();
        // stop and close the processor
        mediaProcessor.stop();
        mediaProcessor.close();
     * Sets the data source. This is where the transmitter will get the media
     * to transmit.
    public void setDataSource(DataSource ds) throws IOException,
        NoProcessorException, CannotRealizeException, NoDataSinkException {
        /* Create the realized processor. By calling the
           createRealizedProcessor() method on the manager, we are guaranteed
           that the processor is both configured and realized already.
           For this reason, this method will block until both of these
           conditions are true. In general, the processor is responsible
           for reading the file from a file and converting it to
           an RTP stream.
        mediaProcessor = Manager.createRealizedProcessor(
            new ProcessorModel(ds, FORMATS, CONTENT_DESCRIPTOR));
        /* Create the data sink. The data sink is used to do the actual work
           of broadcasting the RTP data over a network.
        dataSink = Manager.createDataSink(mediaProcessor.getDataOutput(),
                                          mediaLocator);
     * Prints a usage message to System.out for how to use this class
     * through the command line.
    public static void printUsage() {
        System.out.println("Usage: java MediaTransmitter mediaLocator " +
                           "mediaFile");
        System.out.println(" example: java MediaTransmitter " +
            "rtp://192.168.1.72:49150/audio mysong.mp3");
        System.out.println(" example: java MediaTransmitter " +
            "rtp://192.168.1.255:49150/audio mysong.mp3");
     * Allows the user to run the class through the command line.
     * Only two arguments are allowed; these are the output media
     * locator and the mp3 audio file which will be broadcast
     * in the order.
    public static void main(String[] args) {
        try {
            if (args.length == 2) {
                MediaLocator locator = new MediaLocator(args[0]);
                MediaTransmitter transmitter = new MediaTransmitter(locator);
                System.out.println("-> Created media locator: '" +
                                   locator + "'");
                /* Creates and uses a file reference for the audio file,
                   if a url or any other reference is desired, then this
                   line needs to change.
                File mediaFile = new File(args[1]);
                DataSource source = Manager.createDataSource(
                    new MediaLocator(mediaFile.toURL()));
                System.out.println("-> Created data source: '" +
                                   mediaFile.getAbsolutePath() + "'");
                // set the data source.
                transmitter.setDataSource(source);
                System.out.println("-> Set the data source on the transmitter");
                // start transmitting the file over the network.
                transmitter.startTransmitting();
                System.out.println("-> Transmitting...");
                System.out.println("   Press the Enter key to exit");
                // wait for the user to press Enter to proceed and exit.
                System.in.read();
                System.out.println("-> Exiting");
                transmitter.stopTransmitting();
            } else {
                printUsage();
        } catch (Throwable t) {
            t.printStackTrace();
        System.exit(0);

Okay, here's the it copied out.
Media Transmitter
C:\John\Masters Project\Java\jmf1\MediaPlayer>java MediaTransmitter rtp://127.0.
0.1:2000/audio it-came-upon.mp3
-> Created media locator: 'rtp://127.0.0.1:2000/audio'
-> Created data source: 'C:\John\Masters Project\Java\jmf1\MediaPlayer\it-came-u
pon.mp3'
streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@1decd
ec : 1
sink: setOutputLocator rtp://127.0.0.1:2000/audio
-> Set the data source on the transmitter
-> Transmitting...
   Press the Enter key to exit
MediaPlayerFrame
C:\John\Masters Project\Java\jmf1\MediaPlayer>java MediaPlayerFrame rtp://127.0.
0.1:2000/audio
MediaPlayerFrame
setMediaLocator: rtp://127.0.0.1:2000/audioAs I said, it just kinda stops there, what it should be doing is opening the MediaPlayer.
"MediaPlayerFrame" and "setMediaLocator: rtp://127.0.0.1:2000/audio" are just print outs I used to track here the code is getting to.

DTMF not recognized on connected call

I am not getting any DTMF accptance when I call into an IVR and asked to make a selection 1-9 ect...
ITSP ---->(SIP) CUBE --->(H323) CUCM
pbxguy_router#sho run
Building configuration...
Current configuration : 8466 bytes
! Last configuration change at 23:31:47 UTC Wed Dec 11 2013 by scott
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname pbxguy_router
boot-start-marker
boot system flash c3825-adventerprisek9-mz.151-4.M7.bin
boot-end-marker
no logging buffered
enable secret 5 $1$jLw/$mT4zlcPRsWnipVZ0aHnBA0
aaa new-model
aaa authentication login default local
aaa authentication login sslvpn local
aaa authorization exec default local
aaa session-id common
dot11 syslog
ip source-route
ip cef
ip domain name pbxguy.com
ip name-server 8.8.8.8
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
no ip address trusted authenticate
media flow-around
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
early-offer forced
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
voice translation-rule 1
rule 1 /4029881010/ /1010/
voice translation-rule 2
rule 2 /335201/ /1010/
voice translation-profile INCOMING
translate called 1
voice translation-profile TempIncoming
translate called 2
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1860214740
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1860214740
revocation-check none
rsakeypair TP-self-signed-1860214740
crypto pki certificate chain TP-self-signed-1860214740
certificate self-signed 01
3082022B 30820194 A0030201 02020101 300D0609 2A864886 F70D0101 05050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 31383630 32313437 3430301E 170D3133 31303235 31353434
33375A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 38363032
31343734 3030819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
81009C49 FF6C8071 A0CF9255 70366C11 EDEDE473 013AC5E8 E26B8F1B 5B4FB0CE
B801F441 67439BC7 2A251DD5 708A7A35 128E2DFC 861D23F4 18A23EF0 6AC87CB3
C3F03C47 4285FC3E A043A66B 1232F885 3597F6D6 E98B5E3D 86902237 A3483ED5
17E5B804 A478EAB3 D00D3DEA 68D13EAE 6D9552F3 780E6CB8 B329EEDD 255A22E4
A7610203 010001A3 53305130 0F060355 1D130101 FF040530 030101FF 301F0603
551D2304 18301680 140840B0 1EF7F1E2 1A8CA935 431D067E 57B6C46F 37301D06
03551D0E 04160414 0840B01E F7F1E21A 8CA93543 1D067E57 B6C46F37 300D0609
2A864886 F70D0101 05050003 8181005D 03A541D5 9B3C206E 8BC2E4A3 B00017FE
EC0A4806 5B5E2F3E 67CCDAC9 D11AD33D BD44989F 295E784D E4CF39AC 2E21A2B5
FFAC5171 1372DD0B 764DD3C0 E4088CB7 01D5D4E2 4CA0C955 25F4FF2E 75C3D740
399F67B5 9160F8F4 59206DDC 8392D4B7 47B8E683 220E06BD 2964EBA4 5B57BC98
D623EFC5 399AA46D 6E591D52 45233C
            quit
license udi pid CISCO3825 sn FTX1239A28R
license accept end user agreement
archive
log config
hidekeys
username scott privilege 15 secret 5 $1$Y4nT$1neWA5OmFc/IAvSfA4dB11
redundancy
ip ssh version 2
interface GigabitEthernet0/0
ip address dhcp
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1.40
encapsulation dot1Q 40
ip address 10.10.10.1 255.255.255.252
ip nat inside
ip virtual-reassembly in
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.10.10.1
router eigrp 10
network 10.10.10.1 0.0.0.0
ip local pool webvpn-pool 10.20.30.10 10.20.30.15
ip forward-protocol nd
ip http server
ip http secure-server
ip nat inside source list 1 interface GigabitEthernet0/0 overload
ip nat inside source static udp 192.168.192.3 3074 interface GigabitEthernet0/0 3074
ip nat inside source static udp 192.168.192.3 88 interface GigabitEthernet0/0 88
ip nat inside source static tcp 192.168.192.3 53 interface GigabitEthernet0/0 53
ip nat inside source static udp 192.168.192.3 53 interface GigabitEthernet0/0 53
ip nat inside source static tcp 192.168.192.3 80 interface GigabitEthernet0/0 80
ip nat inside source static tcp 192.168.192.3 3074 interface GigabitEthernet0/0 3074
ip nat inside source static tcp 192.168.254.10 5060 interface GigabitEthernet0/0 5060
ip nat inside source static udp 192.168.254.10 5060 interface GigabitEthernet0/0 5060
ip access-list extended inbound
permit udp any any eq 3074
permit udp any eq 88 any
permit tcp any any eq 3074
permit tcp any any eq domain
permit udp any any eq domain
permit tcp any any eq www
permit tcp any any eq 5060
permit udp any any eq 5060
access-list 1 permit 208.110.65.18
access-list 1 permit 192.168.192.0 0.0.0.255
access-list 1 permit 192.168.128.0 0.0.0.31
access-list 1 permit 10.20.30.0 0.0.0.255
access-list 1 permit 192.168.254.0 0.0.0.255
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.254.10 identifier 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confLab
associate profile 2 register xcodeLab
associate profile 3 register mtpLab
dspfarm profile 2 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 9
associate application SCCP
dspfarm profile 1 conference
codec g711ulaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 1
associate application SCCP
dspfarm profile 3 mtp
codec g711ulaw
maximum sessions hardware 12
maximum sessions software 200
associate application SCCP
dial-peer voice 1 voip
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte h245-signal h245-alphanumeric
no vad
dial-peer voice 2 voip
destination-pattern 4029881010
session target ipv4:192.168.254.10
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 4 voip
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target sip-server
session transport udp
no voice-class sip early-offer forced
dtmf-relay rtp-nte
no dtmf-interworking
codec g711ulaw
no vad
no supplementary-service sip refer
dial-peer voice 5 voip
destination-pattern [2-9]..[2-9]......
session protocol sipv2
session target sip-server
voice-class codec 1
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 335201 password 7 03570B58505E771C165D40 realm sip-ua.com
authentication username 335201 password 7 075C711F18584F554F4652 realm sip-ua.com
registrar dns:proxy.sip-ua.com expires 60
sip-server dns:proxy.sip-ua.com
line con 0
line aux 0
line vty 0 4
exec-timeout 40 0
privilege level 15
password cisco
transport input ssh
scheduler allocate 20000 1000
webvpn gateway Cisco-WebVPN-Gateway
ip address 174.71.48.163 port 443
ssl encryption rc4-md5
ssl trustpoint TP-self-signed-1860214740
logging enable
inservice
webvpn gateway webvpn
ssl trustpoint TP-self-signed-1860214740
no inservice
webvpn install svc flash:/webvpn/anyconnect-win-3.0.11042-k9.pkg sequence 1
webvpn install svc flash:/webvpn/anyconnect-macosx-i386-3.1.04072-k9.pkg sequence 2
webvpn context Cisc0-WebVPN
ssl authenticate verify all
policy group webvpnpolicy
   functions svc-enabled
no inservice
webvpn context Cisco-WebVPN
title "Scott Glenn Private Network, Authorized Users Only"
ssl authenticate verify all
url-list "rewrite"
acl "ssl-acl"
   permit ip 10.20.30.0 255.255.255.0 192.168.192.0 255.255.255.0
   permit ip 10.20.30.0 255.255.255.0 192.168.254.0 255.255.255.0
   permit ip 10.20.30.0 255.255.255.0 10.10.10.0 255.255.255.0
login-message "Cisco Secure WebVPN"
policy group webvpnpolicy
   functions svc-enabled
   svc address-pool "webvpn-pool" netmask 255.255.255.0
   svc rekey method new-tunnel
   svc split include 10.20.30.0 255.255.255.0
   svc split include 192.168.192.0 255.255.255.0
   svc split include 192.168.254.0 255.255.255.0
   svc split include 10.10.10.0 255.255.255.0
default-group-policy webvpnpolicy
aaa authentication list sslvpn
gateway Cisco-WebVPN-Gateway
max-users 2
inservice
end

It appears that I am getting back the following messge when I call into a TFN and try to press option "1":
*Dec 16 14:58:48.604: //20386/xxxxxxxxxxxx/CCAPI/ccCallModifyExtended:
   Nominator=0x716C46E8, Params=0x716C46C0, Call Id=20386
*Dec 16 14:58:48.604: //20387/xxxxxxxxxxxx/CCAPI/ccCallModifyExtended:
   Nominator=0x716C46E8, Params=0x716C46C0, Call Id=20387
*Dec 16 14:58:48.604: //20387/xxxxxxxxxxxx/CCAPI/ccCallModifyExtended:
   Nominator=0x716C46E8, Params=0x716C46C0, Call Id=20387
*Dec 16 14:58:48.664: //20386/xxxxxxxxxxxx/CCAPI/ccCallModify:
   Nominator=0x18E00, Params=0x716C4A50, Call Id=20386
*Dec 16 14:58:48.668: //20386/xxxxxxxxxxxx/CCAPI/cc_api_call_modify_done:
   Result=0, Interface=0x6A030830, Call Id=20386
*Dec 16 14:58:48.668: //20386/xxxxxxxxxxxx/CCAPI/ccCallModifyExtended:
   Nominator=0x716C46E8, Params=0x716C46C0, Call Id=20386
*Dec 16 14:59:01.148: //20384/8008FC7E1300/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x4FA1
*Dec 16 14:59:01.148: //20384/8008FC7E1300/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
*Dec 16 14:59:01.148: //20384/8008FC7E1300/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x4FA1
*Dec 16 14:59:01.148: //20384/8008FC7E1300/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
*Dec 16 14:59:11.280: //20386/xxxxxxxxxxxx/CCAPI/ccCallModify:
   Nominator=0x800, Params=0x716C4A98, Call Id=20386

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
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