[pjsip] Symbian - SIP call breaks on Answer/Ack

Lalit Manchanda Lalit.Manchanda at hsc.com
Wed Sep 12 11:46:01 EDT 2007


Hi Benny,

We are able to run symbian UA on S60 device using WiFi network. When we 
place a call to Symbian-UA from desktop-UA( PjSip)  and it answers the 
call then on receving ACK from destop-UA it terminates the call saying:

 20:21:24.456    dlg0x7136fc Received Request msg ACK/cseq=1869813823 
(rdata0x706d24)
 20:39:59.484    tsx0x713f04 Request to terminate transaction
 20:39:59.513    tsx0x713f04 State changed from Completed to Terminated, 
event=USER
 20:39:59.542    dlg0x7136fc Transaction tsx0x713f04 state changed to 
Terminated
 20:39:59.573 symbian_ua.cpp Call 0 state=CONFIRMED
 20:39:59.629   strm0x717074 RTP status: badpt=0, badssrc=0, dup=0, 
outorder=-1, probation=0, restart=0
 20:39:59.690   strm0x717074 RTP status: badpt=0, badssrc=0, dup=0, 
outorder=0, probation=-1, restart=-1
 20:39:59.760   strm0x717074 Jitter buffer reset
 20:40:00.025    tsx0x713f04 Retransmit timer event
 20:40:00.046    tsx0x713f04 State changed from Terminated to Destroyed, 
event=TIMER
 20:40:00.076   tdta0x7159d4 Destroying txdata Response msg 
200/INVITE/cseq=1869813823 (tdta0x7159d4)
 20:40:00.115    tsx0x713f04 Transaction destroyed! 



I have attached the call logs (of symbian-UA) and ethreal logs of the call 
in incoming_call.zip.

When we try place a call from Symbian-ua to desktop-ua, then on receving 
200 OK, the call is terminated saying:

0:49:55.677    tsx0x713c24 Incoming Response msg 
200/INVITE/cseq=1474304190 (rdata0x706d24) in state Proceeding
 20:49:55.717    tsx0x713c24 State changed from Proceeding to Terminated, 
event=RX_MSG
 20:49:55.747    dlg0x70c064 Received Response msg 
200/INVITE/cseq=1474304190 (rdata0x706d24)
 20:49:55.786    dlg0x70c064 Transaction tsx0x713c24 state changed to 
Terminated
 20:49:55.815 symbian_ua.cpp Call 0 state=CONNECTING
 20:49:55.846    inv0x70c064 Got SDP answer in Response msg 
200/INVITE/cseq=1474304190 (rdata0x706d24)
 20:49:55.885    inv0x70c064 SDP negotiation done, status=0
 20:49:55.937  icstr0x707cb8 Disabling local ICE, reason=ice-ufrag 
attribute not found.

The call logs and ethreal logs can be seen in outgoing_call.zip


Kindly give some hints, to take it forward

Lalit Manchanda







"Benny Prijono" <bennylp at pjsip.org> 
Sent by: pjsip-bounces at lists.pjsip.org
09/11/2007 09:25 PM
Please respond to
Benny Prijono <bennylp at pjsip.org>; Please respond to
pjsip embedded/DSP SIP discussion <pjsip at lists.pjsip.org>


To
pjsip embedded/DSP SIP discussion <pjsip at lists.pjsip.org>
cc

Subject
Re: [pjsip] playsine does not play on symbian






Cool! At least now I know that it will really play sounds.

We can't play full duplex because Nokia says so. Full duplex is only 
accessible with some sort of special license from Nokia.

I forgot why we need CActiveSchedulerWait. I concluded that from reading 
Symbian docs.

--
Benny Prijono
http://www.pjsip.org


- original message -
Subject:                 Re: [pjsip] playsine does not play on symbian
From:            Lalit Manchanda <Lalit.Manchanda at hsc.com>
Date:                            11/09/2007 11:13

Hi Benny,

Thanks a lot for quick and help ful reply. We here also were able to get 
events now, to do the same I start a nested Schedular in 
CPjAudioOutputEngine::StartPlay (Symbian_sound.cpp), creating 
CActiveSchedulerWait as class variable and then starting it. Were able to 
play sound by providing a providing a wav file as buffer.

So after this symbian ua menu is showing up on device!  Why do you think 
we can't do full duplex? 

I have a question regarding use of CActiveSchedulerWait  in 
main_symbian.cpp, Did you have any specific use which installed 
ActiveScheduler won't do?

Regds
Lalit Manchanda




Benny Prijono <bennylp at pjsip.org> 
Sent by: pjsip-bounces at lists.pjsip.org
09/10/2007 08:12 PM
Please respond to
pjsip embedded/DSP SIP discussion <pjsip at lists.pjsip.org>


To
pjsip at pjsip.org
cc

Subject
Re: [pjsip] playsine does not play on symbian






Lalit Manchanda wrote:
> 
> Hi Benny,
> 
> I have used the same framework you created for symbian_ua, so I think 
> Active Scheduler is installed, though none of the callbacks get called. 
> I have checked various forums it seems to be Active Schduler problem. I 
> am trying to dig into it, Please share ideas if any

I did some work into Symbian sound couple of weeks back, and at
least I saw the callbacks being called (after some bug fixing, of
course!).

Check out the latest SVN trunk, it now has a separate sound test
program for Symbian (symsndtest.mmp). I haven't tested anything like
playing sounds yet, so it may still not play sine waves, but maybe
you'll get better result. But one thing for sure, full duplex audio
doesn't work on S60 3rd edition.

  -benny



_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip at lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org





*****************************************************DISCLAIMER*****************************************************

This message and/or attachment(s) contained here are confidential, 
proprietary to HUGHES SYSTIQUE and its customers. 
Contents may be privileged or otherwise protected by law. The information 
is solely intended for the entity it is 
addressed to. If you are not the intended recipient of this message, it is 
strictly prohibited to read, forward, 
print, retain, copy or disseminate this message or any part of it. If you 
have received this e-mail in error, 
please notify the sender immediately and delete the message.

********************************************************************************************************************


_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip at lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org


_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip at lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org





*****************************************************DISCLAIMER*****************************************************

This message and/or attachment(s) contained here are confidential, proprietary to HUGHES SYSTIQUE and its customers. 
Contents may be privileged or otherwise protected by law. The information is solely intended for the entity it is 
addressed to. If you are not the intended recipient of this message, it is strictly prohibited to read, forward, 
print, retain, copy or disseminate this message or any part of it. If you have received this e-mail in error, 
please notify the sender immediately and delete the message.

********************************************************************************************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20070912/1ea702ee/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: outgoing_call.zip
Type: application/zip
Size: 8065 bytes
Desc: not available
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20070912/1ea702ee/attachment.zip>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: incoming_call.zip
Type: application/zip
Size: 8618 bytes
Desc: not available
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20070912/1ea702ee/attachment-0001.zip>


More information about the pjsip mailing list