Hi All, We were able to establish a call between two Windows Mobile handsets. But sound is audible only on one device. The scenario is as follows: If there are two devices D1 and D2, and a call is made from D1 to D2, it is audible only on D2. Even if call is made from D2 to D1, the sound is still audible on D2 side with very poor quality over wifi/GPRS. We have tried to set the following parameters also, but it has only increased the sound intensity keeping the choppiness of sound and packet loss intact on D2. PJMEDIA_SND_DEFAULT_REC_LATENCY set to 140 PJMEDIA_SND_DEFAULT_PLAY_LATENCY set to 140 We have found that both the handsets were using differnt codecs as displayed by wireshark in RTP packets. Those were G711 PCMU and GSM codecs. 1) Is it necessary to have same codec between two communicating devices? 2) What is the ideal configuration settings for WM devices? 3)How can we determine CPU usage while our application is running as it can be one of the reasons behind packet loss. Please note that we are using PJSIP version 1.5 The handsets we are using are HTC Touch(D1) and HTC innovation(D2). We have attached the PJSIP.log file with this mail. Thanks, Sangram -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091216/28da5fe0/attachment-0001.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: pjsip.log Type: application/octet-stream Size: 399859 bytes Desc: not available URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091216/28da5fe0/attachment-0001.log>
