Hi,
Thanks everyone for helping out. I managed to make it work. I don’t have a
proxy server in the middle. I was using the sipp for the RTP stream portion for
some media related project.
My uac_pcap.xml was a strict copy from the embedded uac_pcap. However I changed
“c=IN IP[local_ip_type] [local_ip]” to “c=IN IP[media_ip_type] [media_ip]” , on
the client side, the command issued is:
./sipp –sf uac_pcap.xml 192.168.1.237 –mi 192.168.1.236
Where 192.168.1.236 is the client side ip address and 192.168.1.237 is the
server IP address
On the server side, my uas.xml is a strict copy from the embedded uas, the
command issued is:
./sipp –sf uas.xml –mi 192.168.1.237
With the above change and commands, I was able to get both signaling and rtp
working.
I hope sipp masters can document a RTP stream example for client/server on
different machines. That would help new comers a lot.
Thanks,
Sherry
From: Kenneth [mailto:chengku...@gmail.com]
Sent: Tuesday, September 30, 2014 2:05 AM
To: Sherry Wei
Cc: Sakharam Thorat; sipp-users
Subject: Re: [Sipp-users] uac_pcap sending RTP packets to localhost 127.0.0.1
Yes on lo interface you can see the rtp streams in/out of 127.0.0.1
I guess you use no sip proxy like asterisk. Someone has already posted you a
solution. Try that first. If that doesn't work, please email me and I will send
you my scripts. Currently I am in a LAN test so I cannot visit gmail on my
computer.
This email is typed on my iPhone. I'd like to apologize for any mistake in it.
在 2014年9月30日，1:01，"Sherry Wei" <s...@carmelosystems.com
<mailto:s...@carmelosystems.com> > 写道：
Hi Sakharam and Cheng,
I am using the sudo to issue the command. Here is the xml script I use which is
exactly a copy from the embedded scenario uac_pcap (I did a –sd to dump the
embedded uac_pcap scenario to uac_pcap.xml file). Anything wrong in this
script? Also, at the end, is the tcpdump that on lo interface that shows all
rtp packets send and rcv on 127.0.0.1
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[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 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause 8 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="8000"/>
<!-- Play an out of band DTMF '1' -->
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
Tcpump –I lo udp –n –xxx
09:59:07.414542 00:00:00:00:00:00 > 00:00:00:00:00:00, ethertype IPv4 (0x0800),
length 294: 127.0.0.1.6144 > 127.0.1.1.6000: UDP, length 252
0x0000: 0000 0000 0000 0000 0000 0000 0800 4500
0x0010: 0118 0000 4000 4011 3ad3 7f00 0001 7f00
0x0020: 0101 1800 1770 0104 f8bd 8008 e718 0000
0x0030: 1a40 dee0 ee8f 5a5a 5a5a 5ad5 5a5a 5a5a
0x0040: 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a
0x0050: 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a
0x0060: 5a5a d55a 5a5a 5a5a 5a5a 5a5a 5a5a d55a
0x0070: 5a5a 5a5a 5ad5 5a5a 5a5a 5a5a d5d5 d55a
0x0080: d5d5 d5d5 d55a d5d5 d5d5 d5d5 d5d5 d55a
0x0090: d55a d5d5 d5d5 d5d5 d5d5 d5d5 d5d5 d5d5
0x00a0: d5d5 d55a 5ad5 d5d5 5a5a d5d5 d5d5 d5d5
0x00b0: d5d5 d5d5 d5d5 d5d5 5a5a d5d5 5ad5 d5d5
0x00c0: 5a5a d55a 5ad5 5a5a 5a5a 5a5a 5a5a 5a5a
0x00d0: 5a5a 5a5a 5a5a 5a5a 5a5a 5ad5 5a5a 5a5a
0x00e0: 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a
0x00f0: 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a
0x0100: 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a
0x0110: 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a 5a5a
0x0120: 5a5a 5a5a 5a5a
From: Sakharam Thorat [mailto:sakharam.tho...@outlook.com]
Sent: Monday, September 29, 2014 6:19 AM
To: Sherry Wei
Cc: sipp-users
Subject: RE: [Sipp-users] uac_pcap sending RTP packets to localhost 127.0.0.1
Post sipp scenario, or please cross check following headers in SDP or post
wireshark trace.
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
c=IN IP[local_ip_type] [local_ip}
m=audio [auto_media_port] RTP/AVP 8 101
Best Regards,
Sakharam Thorat.
_____
From: s...@carmelosystems.com <mailto:s...@carmelosystems.com>
To: sipp-users@lists.sourceforge.net <mailto:sipp-users@lists.sourceforge.net>
Date: Sun, 28 Sep 2014 15:44:28 -0700
Subject: [Sipp-users] uac_pcap sending RTP packets to localhost 127.0.0.1
Hi,
I am using sip-3.3 and run into this basic problem.
I ran “./sipp –sn uas” on server 192.168.1.236, and
I ran “./sipp –sn uac_pcap 192.168.1.236” on another machine (192.168.1.237) on
the same subnet.
I can see SIP packets arriving on the server machine, but RTP packets were all
sent to 127.0.0.1 on the client machine!
What did I miss? This sounds like a very basic scenario.
Thanks so much,
Sherry
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