Issue when SRTP enabled in the SIP clients - sr-users

21 Dec 2011

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