Hello All, I am running an IP telephony application on a cellphone. The application uses the pjsip stack. I find that when i place a call using this cellphone application, the call goes through but voice is one sided. The cellphone running the app cannot hear anything. When i look at the log on the cellphone, i see the following error "RTP decode error: Unknown pjmedia error 220122" I looked at the source code and found that in file rtp.c, there is a check for version number in the RTP header. /* Check RTP header sanity. */ if ((*hdr)->v != RTP_VERSION) { return PJMEDIA_RTP_EINVER; } and RTP_VERSION is defined as 2. Using wireshark, i have captured RTP packets destined to the cellphone and upon examining the RTP header i see that the version number is indeed 2. What else could be causing this error ? Is there any other check which would produce this error output upon failure ? I have attached a file showing my wireshark capture. Thanks and Regards, Vikram. PS : I briefly outline the system setup Cellphone uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Router is connected to a modem through a series of switches. The cellphone is behind a double NAT. Cellphone app, tries to register with Asterisk through an outbound proxy server(Kamailio + rtpproxy). The proxy server rewrites SDP to force rtp through an rtpproxy. Both Asterisk and and the outbound proxy have Public IP addresses. -------------- next part -------------- A non-text attachment was scrubbed... Name: RTP_packet_capture.jpg Type: image/jpeg Size: 133147 bytes Desc: not available URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100215/2885f3d5/attachment-0001.jpg>