Hi guys, I've been trying to make a sip server using PJSUA, and using linphone and vimphone clients to test it. After some efforts, by modifying the messages on_rx_data and on_tx_data i could pass the sip signalling messages properly But after the states going to "confirmed" state, i could hear audio (bidirectional) only if client (using stunserver) makes a call to server. When server tries to make a call to the registrar fetched address of the client, i find only upbandwidth in client but no downbandwidth. Auto RTP switching feature ( PJMEDIA_UDP_NO_SRC_ADDR_CHECKING) is turned on (set to 0) in PJMEDIA_TRANSPORT_UDP_ATTACH api How to overcome this?? Please help!! Thank u, Praveen -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140529/b21937d5/attachment.html>
