Hi Fabio, in the project "pjproject-1.0.1" the Symbian project file (mmp) has defined only 2 codecs: GSM and Speex. I added in my project the "iLBC" codec. What I understood is during the SDP exchange the SIP phone is sending in its media description the used codec. I think they are negotiating the codec usage but I don't know who decide which is the codec to use (the Phone client or the server). The last tests I made are with a Nokia E61 connected in Wi-Fi. I'm new in "pjlib" but as you suggested to look for buffer overflowing or underflowing, where should I look? In jitter buffer? Thanks, George. _____ From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org] On Behalf Of Fabio Pietrosanti (naif) Sent: Thursday, March 19, 2009 1:16 PM To: pjsip list Subject: Re: Pjlib on Symbian Did you tried different codes? Like GSM, G729, AMR, G711? Please consider that the 3G connection and the 'packetizing algorithm' has to be carefully design on Symbian in order not to incurr in overflow and underflow. Did you checked if there's some buffer overflowing or underflowing? Fabio George Evi wrote: Hi Benny, Thanks for your response. The flow of sound is disrupted on both sides (caller and callee voice reception). You can hear the sound but the words are not completed and on the callee side the voice is metalique (like a robot speech). The latest tests I made were done with Nokia E61 (S60 3rd edition -mr) connected Wi-Fi and I expected to see some improvements but voice stilled disrupted. I'm using iLBC as codec (1st priority) and UDP transport. Do you have any suggestions? Thank you, George. _____ From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org] On Behalf Of Benny Prijono Sent: Thursday, March 19, 2009 11:27 AM To: pjsip list Subject: Re: Pjlib on Symbian 2009/3/17 George Evi <george.evi at ctcinc.ca> Hi, I created an application based on pjlib on Symbian OS and have some problems with the sound. The SIP session is OK but during the media session the sound is interrupted at the caller side and the destination has an echo (during of all conversation) and the sound is interrupted also. By "interupted", did you mean like stuttering? Does it happen often? If it is stutter, it could be caused by network jitter, or some activity in the application (we found that even simple activity such as printing log message to console screen could delay the audio). Regarding the echo, I realize that the echo suppressor in pjmedia is still work in progress, so "a bit" of echo is quite expected. I did the tests with Nokia E71 on 3G network. Do we need a 3G network? I would say Wi-Fi would work better. What are the minimum requirements (Phone hardware: processor, memory and others) to have a good quality of sound? Minimum requirement is an S60 3rd ed device. For best quality, use APS-Direct ([1], to be included in release 1.1). It uses native/handset's codec and echo canceller, and in my personal test there is zero echo with this. Though the drawback is it needs Symbian signing. cheers Benny [1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct Thanks, George. _______________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090319/5fd8dbb7/attachment.html>
