[sip-comm-dev] html reports


#1

Hello all,

Just a line to let you know that I've added a new target (which should remain hidden I guess) in the build.xml. The target generates html reports from the xml file that has resulted from running the test target. Html reports are stored in the test-reports/html directory and are automatically generated at the end of the test target.

I've been using them for a while and find them quite useful.

Cheers
Emil

···

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#2

Hello all, I saw a new version of the project in the java.net. This version had a executable for windows in Java Web Start and the project logo was different too.
But when I look another day the new version wasn't more in java.net.
Anyone can tell me what happened ?

[]s

Emil Ivov <emil.ivov@gmail.com> escreveu: Hello all,

Just a line to let you know that I've added a new target (which should
remain hidden I guess) in the build.xml. The target generates html
reports from the xml file that has resulted from running the test
target. Html reports are stored in the test-reports/html directory and
are automatically generated at the end of the test target.

I've been using them for a while and find them quite useful.

Cheers
Emil

···

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Atenciosamente, Lucas Bittencourt Xavier.

---------------------------------
Yahoo! doce lar. Fa�a do Yahoo! sua homepage.


#3

Hello Lucas,

Lucas Bittencourt wrote:

Hello all, I saw a new version of the project in the java.net. This version had a executable for windows in Java Web Start and the project logo was different too.
But when I look another day the new version wasn't more in java.net.
Anyone can tell me what happened ?

As you noticed, we're currently working on a new (1.0) version of the SIP Communicator. Previous versions are what we now call "pre 1.0 version" and are discontinued.

This major change in the core of the SIP Communicator came with a new licence (from Apache to LGPL), a new logo and a new website located at http://www.sip-communicator.org. Browsing this one may answer most of your questions, however don't hesitate to post here if you still have some.

Cheers,
Martin

···

[]s

*/Emil Ivov <emil.ivov@gmail.com>/* escreveu:

    Hello all,

    Just a line to let you know that I've added a new target (which should
    remain hidden I guess) in the build.xml. The target generates html
    reports from the xml file that has resulted from running the test
    target. Html reports are stored in the test-reports/html directory and
    are automatically generated at the end of the test target.

    I've been using them for a while and find them quite useful.

    Cheers
    Emil

    ---------------------------------------------------------------------
    To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
    For additional commands, e-mail: dev-help@sip-communicator.dev.java.net

Atenciosamente, Lucas Bittencourt Xavier.

------------------------------------------------------------------------
Yahoo! doce lar. Fa�a do Yahoo! sua homepage. <http://us.rd.yahoo.com/mail/br/tagline/homepage_set/*http://br.yahoo.com/homepageset.html>

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#4

Hi all!

I was wondering if anyone else has stumled upon this issue and maybe has a fix for it. When I receive DTMF signals (according to RFC 2833) from the remote party of my call, that audio channels dies for me in Sip Communicator. That is, I can't hear the other party anymore.

I see an error message on the console saying "No format has been registered for RTP Payload type 101". I have also traced this error a bit and it seems that in AVReceiver.java events RemotePayloadChangeEvent is received and after that ControllerErrorEvent. Probably because the RTPManager and JMF can't handle the new RTP stream. I have tried to register my dummy codec for that particular payload and the error message disappears but still the audio channels dies on my.

Has anyone else got this error or wondered about this quite serious DTMF problem in SipComm before? :slight_smile:

Thanks in advance!

Kind regards,

Johan Paul

···

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#5

Hi all!

Hi Johan,

I was wondering if anyone else has stumled upon this issue and maybe has
a fix for it. When I receive DTMF signals (according to RFC 2833) from
the remote party of my call, that audio channels dies for me in Sip
Communicator. That is, I can't hear the other party anymore.

I see an error message on the console saying "No format has been
registered for RTP Payload type 101". I have also traced this error a

you should restart AVReceiver after RTP DTMF. I didn't find other way.

it's not so difficult :slight_smile:

···

On Mon, Feb 06, 2006 at 08:54:53AM +0200, Johan Paul wrote:

bit and it seems that in AVReceiver.java events RemotePayloadChangeEvent
is received and after that ControllerErrorEvent. Probably because the
RTPManager and JMF can't handle the new RTP stream. I have tried to
register my dummy codec for that particular payload and the error
message disappears but still the audio channels dies on my.

Has anyone else got this error or wondered about this quite serious DTMF
problem in SipComm before? :slight_smile:

Thanks in advance!

Kind regards,

Johan Paul

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#6

I implemented DTMF using SIP dtmf relay.

http://www.voip-info.org/wiki/view/SIP+Info+DTMF

Cheers!
lumwaiph

···

On 2/6/06, Alexandr Dubovikov <shurik@start4.info> wrote:

On Mon, Feb 06, 2006 at 08:54:53AM +0200, Johan Paul wrote:
> Hi all!

Hi Johan,

>
> I was wondering if anyone else has stumled upon this issue and maybe has
> a fix for it. When I receive DTMF signals (according to RFC 2833) from
> the remote party of my call, that audio channels dies for me in Sip
> Communicator. That is, I can't hear the other party anymore.
>
> I see an error message on the console saying "No format has been
> registered for RTP Payload type 101". I have also traced this error a

you should restart AVReceiver after RTP DTMF. I didn't find other way.

it's not so difficult :slight_smile:

> bit and it seems that in AVReceiver.java events RemotePayloadChangeEvent
> is received and after that ControllerErrorEvent. Probably because the
> RTPManager and JMF can't handle the new RTP stream. I have tried to
> register my dummy codec for that particular payload and the error
> message disappears but still the audio channels dies on my.
>
> Has anyone else got this error or wondered about this quite serious DTMF
> problem in SipComm before? :slight_smile:
>
> Thanks in advance!
>
>
> Kind regards,
>
> Johan Paul
>
> ---------------------------------------------------------------------
> To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
> For additional commands, e-mail: dev-help@sip-communicator.dev.java.net
>

--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
    AD1-UANIC * ICQ: 122351182 * http://www.start4.info

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#7

Hi Wai!

Thanks for your post! Interesting detail, I shall put it behind my ear!

But unfortunately my problem was not sending DTMF, but receiving it in
RFC specified format, which most of the SIP clients use as default.

I am trying to restart the AVReceiver by calling initialize() again
after the DTMF RTP packets. Unfortunately it doesn't seem to open the
voice channel again... I have check my code.

You shouldn't call one more time inititalize(), becase it will open new RTP
listener. Try todo the same like I did:

I don't remember that was in original SipCommunicator, but it should be like
this:

protected void restartAll() throws MediaException {

    if (mgrs == null) return;

    if (console.isDebugEnabled()) {
      console.debug("Preparing to restart receiving for:" + callid);
    }

    try {
      for (int i = 0; i < mgrs.length; i++) {
        if (mgrs[i] == null) {
          continue;
        }
        RTPSocketAdapter rtpSocketAdapter = mediaManager.getRtpConnector(callid);
        if(rtpSocketAdapter==null) return;
        mgrs[i].initialize(rtpSocketAdapter);
        if (console.isDebugEnabled()) {
          console.debug("Restart receiving for:" + callid);
        }
        if (console.isDebugEnabled()) {
          console.debug("Started transmitting one more time:" + callid);
        }
      }
    }
    catch (Exception ex) {
      console.error("Session " + callid +
                    " failed to start receiving.");
      throw new MediaException(
          "Session " + callid + " failed to start transmitting.");
    }
  }

···

On Mon, Feb 06, 2006 at 12:55:56PM +0200, Johan Paul wrote:

--
Johan Paul

--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
    AD1-UANIC * ICQ: 122351182 * http://www.start4.info

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#8

Hi Wai!

Thanks for your post! Interesting detail, I shall put it behind my ear!

But unfortunately my problem was not sending DTMF, but receiving it in RFC specified format, which most of the SIP clients use as default.

I am trying to restart the AVReceiver by calling initialize() again after the DTMF RTP packets. Unfortunately it doesn't seem to open the voice channel again... I have check my code.

···

--
Johan Paul

I implemented DTMF using SIP dtmf relay.

http://www.voip-info.org/wiki/view/SIP+Info+DTMF

Cheers!
lumwaiph

On 2/6/06, *Alexandr Dubovikov* <shurik@start4.info > <mailto:shurik@start4.info>> wrote:

    On Mon, Feb 06, 2006 at 08:54:53AM +0200, Johan Paul wrote:
     > Hi all!

    Hi Johan,

     >
     > I was wondering if anyone else has stumled upon this issue and
    maybe has
     > a fix for it. When I receive DTMF signals (according to RFC 2833)
    from
     > the remote party of my call, that audio channels dies for me in Sip
     > Communicator. That is, I can't hear the other party anymore.
     >
     > I see an error message on the console saying "No format has been
     > registered for RTP Payload type 101". I have also traced this error a

    you should restart AVReceiver after RTP DTMF. I didn't find other way.

    it's not so difficult :slight_smile:

     > bit and it seems that in AVReceiver.java events
    RemotePayloadChangeEvent
     > is received and after that ControllerErrorEvent. Probably because the
     > RTPManager and JMF can't handle the new RTP stream. I have tried to
     > register my dummy codec for that particular payload and the error
     > message disappears but still the audio channels dies on my.
     >
     > Has anyone else got this error or wondered about this quite
    serious DTMF
     > problem in SipComm before? :slight_smile:
     >
     > Thanks in advance!
     >
     > Kind regards,
     >
     > Johan Paul
     >
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     > To unsubscribe, e-mail:
    dev-unsubscribe@sip-communicator.dev.java.net
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     > For additional commands, e-mail:
    dev-help@sip-communicator.dev.java.net
    <mailto:dev-help@sip-communicator.dev.java.net>
     >

    --
    Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
    <mailto:shurik@start4.info>
        AD1-UANIC * ICQ: 122351182 * http://www.start4.info

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--
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