Hi All,
I am using Sipp 3.1 on Windows XP to generate multiple calls with RTP (audio
media).
Here are my details to run UAC and UAS
For UAC:
Command line arguments: sipp -sf UacWithOwnMediaMultipleUAC.xml uac -s
dinesh -t un -i 192.168.96.202 192.168.96.203 -r 50 -rp 25000 -m 10000 -l
50 -trace_err
For UAS:
Command line arguments: sipp -sf UasWithMultipleRtpPort_OwnMedia.xml uas
192.168.96.202 -mi 192.168.96.203 -mp 7000 -rtp_echo
When UAC sends RTP packets to UAS only the first call's RTP packets are
echoed back and with rest of calls, UAS sends ICMP packets with respect to
RTP packets send from UAC. When I dig into the issue, I found that even
though UAS sends a unique RTP Port entry every time in SDP with SIP OK, it
does not open that port for Media Exchange.
SIP OK from UasWithMultipleRtpPort_OwnMedia.xml
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
SIP INVITE from UacWithOwnMediaMultipleUAC.xml.
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
Can anyone help me to fix this issue? I am a newbie with Sipp. I want to
generate 200 calls simultaneously with my own RTP audio media.
Regards,
Dinesh
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