Hi All,
I am developing SIP Client using PJSIP Lib. I have two application (one is
UI interface and another is PJSIP lib). My UI application have 2-3 external
sound source which are connected through USB port. My UI application
responsible for sending and receiving audio packets from that devices.
Now i want to send/receive adio frames frop PJSIP lib so that i can direct
them to my sound devices and vice-versa.
I have used set_no_snd_dev() and sending/receiving audio frames using
pjmedia_port_get_frame() , pjmedia_port_put_frame() from PJ SIP Lib.
but using the technique
->Receiving Audio Frames and playing it on my sound device is
working fine.
->Audio is getting stretched on remote end (another SIP client ).
I am using PCMA (8000Hz/Mono channel/160 sample per frame).
When is saw pjsip log, i found "Master/Sound Under flow" errors which are
coming after every 1ms:
Please suggest me , what is i am doing wrong????
Regards
*Vijay Kumar*
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