Hi all. I was wondering if pjsua application manages the case when during a call the RTP stream is interrupted for a certain amount of time (let say 10 seconds)? If It doesn't, does anyone know if there is any SIP related standard (RFC or other) describing a common behaviour to be implemented in a SIP client for this situation. Thank you Massimiliano -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080930/672f25db/attachment-0001.html>
