[sip-comm-dev] Sip Communicator loses sound after pressing a keypad number in IVR (SIP protocol)


#1

Hello,

In recent builds of Sip Communicator o have problems with IVRs (again).

Now it is an interesting issue:

I call the phone number (from SIP account via an Asterisk/Trixbox server), i hear the automated voice recording asking to press 1, 2 etc for different options. I open the dialpad, press a number and all goes silent. The call is not interrupted.
Every time i have 100+ CPU usage in such cases from the beginning of the call. I use Debian Squeeze 32-bit.
This happens only with Sip Communicator, with Sflphone/Ekiga/Linphone it works. My account is set to use sip info by default. The other SIP clients have the option to force either sip info or inband dtmf, Sip Communicator ahould have this option too.

Any thoughts?

···

--
O zi buna,

Kertesz Laszlo

Using Opera's revolutionary email client: http://www.opera.com/mail/

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#2

I still have the issue in build 3213, without the high cpu usage.

It seems that the connection is not lost, because i still have incoming/outgoing 10 KB/s streams only that i cant hear anything.

In the log i have this:

09:58:45.456 WARNING: impl.protocol.sip.SipApplicationData.getApplicationData().96 container is null
09:58:45.659 INFO: impl.neomedia.MediaStreamImpl.printFlowStatistics().2232 rtpstat:call stats for outgoing audio stream SSRC:1861158215
rtpstat:bytes sent: 87520
rtpstat:RTP sent: 547
rtpstat:remote reported min interarrival jitter : 81
rtpstat:remote reported max interarrival jitter : 202
rtpstat:local collisions: 0
rtpstat:remote collisions: 0
rtpstat:RTCP sent: 3
rtpstat:transmit failed: 0
09:58:45.660 INFO: impl.neomedia.MediaStreamImpl.printFlowStatistics().2280 rtpstat:call stats for incoming audio stream SSRC:729501424
rtpstat:packets received: 560
rtpstat:bytes received: 96104
rtpstat:packets lost: 0
rtpstat:min interarrival jitter : 0
rtpstat:max interarrival jitter : 11
rtpstat:RTCPs received: 2
rtpstat:bad RTCP packets: 0
rtpstat:bad RTP packets: 0
rtpstat:local collisions: 0
rtpstat:malformed BYEs: 0
rtpstat:malformed RRs: 0
rtpstat:malformed SDESs: 0
rtpstat:malformed SRs: 0
rtpstat:packets looped: 0
rtpstat:remote collisions: 0
rtpstat:SRRs received: 2
rtpstat:transmit failed: 0
rtpstat:unknown types: 0
09:58:45.662 SEVERE: util.UtilActivator.uncaughtException().81 An uncaught exception occurred in thread=Thread[RTCP Reporter,9,system] and message was: SSRCTable Enumeration
java.util.NoSuchElementException: SSRCTable Enumeration
  at com.sun.media.rtp.util.SSRCTable$1.nextElement(SSRCTable.java:229)
  at com.sun.media.rtp.SSRCCache.aliveCount(SSRCCache.java:211)
  at com.sun.media.rtp.SSRCCache.calcReportInterval(SSRCCache.java:489)
  at com.sun.media.rtp.RTCPReporter.run(RTCPReporter.java:178)
  at java.lang.Thread.run(Thread.java:662)

···

On Wed, 22 Dec 2010 15:47:09 +0200, Kertesz Laszlo <laszlo.kertesz@gmail.com> wrote:

Hello,

In recent builds of Sip Communicator o have problems with IVRs (again).

Now it is an interesting issue:

I call the phone number (from SIP account via an Asterisk/Trixbox server), i hear the automated voice recording asking to press 1, 2 etc for different options. I open the dialpad, press a number and all goes silent. The call is not interrupted.
Every time i have 100+ CPU usage in such cases from the beginning of the call. I use Debian Squeeze 32-bit.
This happens only with Sip Communicator, with Sflphone/Ekiga/Linphone it works. My account is set to use sip info by default. The other SIP clients have the option to force either sip info or inband dtmf, Sip Communicator ahould have this option too.

Any thoughts?

--
O zi buna,

Kertesz Laszlo

Using Opera's revolutionary email client: http://www.opera.com/mail/

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