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Hi,
Is it OK I can ask question here? otherwise could you help to point a place?
I'm using sipp to do a multiple call test, on server side, I noticed it only listen on min_rtp_port. So, call test with RTP only works for 1st call. For the following call, sipp server says it's using "min_rtp_port+2" in "200OK SDP", but, not actually listening on that port. When peer side send rtp heading to that port, system respond with "port unreachable".
Could you tell how to debug this?
oot@admin2-ubu1951:/env_test/env_repository/app_cfg/voice/sipp/call# sipp -v
SIPp v3.7.3-TLS-SCTP-PCAP-SHA256.
command to start sipp server:
oot@admin2-ubu1951:/env_test/env_repository/app_cfg/voice/sipp/call# sipp -sf uas.xml -i 10.25.75.180 -min_rtp_port 10000 -max_rtp_port 40000 -rtp_echo -rtpcheck_debug
Checking listening port, can see it always listen 10000 and 10002. It's not moving when receiving more calls
oot@admin2-ubu1951:/env_test/env_repository/app_cfg/voice/sipp/call# cat uas.xml
Content-Length: 0 ]]> Content-Type: application/sdp Content-Length: [len]v=0
o=user1 53655765 2353687637 IN IP4 [local_ip]
s=-
c=IN IP4 [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
root@admin2-ubu1951:/env_test/env_repository/app_cfg/voice/sipp/call#
Thanks for the help!