Hi, I am also facing the same issue as reported by Steve. Some more observations from my side:- 1. The Sound device callbacks to stream (get_frame and put_frame) are still working after GSM call has been made although it seems that only silence is captured as the VAD drops all the frames and only minimal RTP (marker bit set) packets are sent to other party. 2. There is another strange observation. As Steve mentioned, once a GSM has been done and VoIP call is done thereafter, there is a just one-way voice flow from other party to windows mobile device. Now while this VoIP call is going on, initiate an incoming GSM call to this device. For the duration when the Device is ringing for incoming GSM call, the voice is getting correctly captured from the microphone and 2-way voice flow resumes (although ringing tone adds to captured voice). Can anyone throw some light on what can be causing this problem in mic. Regards Girish ________________________________ From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org] On Behalf Of Steven Beck Sent: Tuesday, November 11, 2008 4:18 AM To: pjsip Subject: Fwd: problem with mic on windows mobile I found a bug in wmme_sound.c, but fixing the bug didn't fix the mic problem. In that file, in pjmedia_snd_init the code needs to zero dev_info before calling WideCharToMultiByte to convert the device name. What is happening now is each time this method is called to init the sound device, the "- Input" gets added to the name, making the name grow each time. But this did not fix the mic problem. To repeat the problem - after the pda gets a cell call, pjsip no longer gets sound from the mic. The only solution I've found so far is to exit and restart my program, which of course is not acceptable. ---------- Forwarded message ---------- From: Steven Beck <sbeck45@xxxxxxxxx<mailto:sbeck45 at gmail.com>> Date: Mon, Nov 10, 2008 at 1:00 PM Subject: Fwd: problem with mic on windows mobile To: pjsip <pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>> I really could use some help here - please ! After getting a cell call, I called pjsua_destroy() followed by re-initializing it (pjsua_create, pjsua_init, pjsua_transport_create and pjsua_start). Even after all of that the microphone (sound from PDA to other side of call) is not working. But if I exit my program and restart my program, then the mic is working again. It seems that something in pjsip isn't correctly re-initialized, but what?? ---------- Forwarded message ---------- From: Steven Beck <sbeck45@xxxxxxxxx<mailto:sbeck45 at gmail.com>> Date: Sun, Nov 9, 2008 at 5:28 PM Subject: problem with mic on windows mobile To: pjsip <pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>> I was playing around with your package (very cool by the way) on windows mobile. Sound in calls seems to be working fine. But if the mobile device gets a cell call, after that, the mic doesn't work in pjsip. Any ideas what's wrong? Is it possible to be notified when pjsip loses the mic? What needs to be done to get the mic back? Steven **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS******** End of Disclaimer ********INFOSYS*** -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20081112/b2aa7c63/attachment.html>
