Hi,

I followed this web article to install Kamailio 3.2 and RTPProxy on Debian Squeeze x64:

http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour

The system is running on a public IP address outside of our corporate LAN.  I have been testing it using pjsua v2 alpha 2 from the pjsip.org project.

I am having an issue when I enable srtp in the pjsua clients.  If both pjsua clients are running on machines on our corporate LAN (symmetric NAT), the call succeeds and I get audio and video.  If one of the clients is running outside of the corporate LAN, the call connects but I do not get any audio or video.  If I turn off srtp in both clients and try the call again, audio and video starts working.  Is there any additional configuration I need to make in the kamailio.cfg file when I intend to use srtp in the clients?

RTPProxy info:
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup command
Extension 20081224: Support for session timeout notifications

Kamailio info:
version: kamailio 3.2.0 (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 10:23:25 Nov  2 2011 with gcc 4.4.5

Regards,
--Jonathan