Hi Sherry,
Can you use -mi (for media IP) and -mp (for media port) in the same command.

Thanks

-Abinash 


From: s...@carmelosystems.com
To: sakharam.tho...@outlook.com
Date: Mon, 29 Sep 2014 10:01:45 -0700
CC: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] uac_pcap sending RTP packets to localhost 127.0.0.1

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
To: sipp-users@lists.sourceforge.net
Date: Sun, 28 Sep 2014 15:44:28 -0700
Subject: [Sipp-users] uac_pcap sending RTP packets to localhost 127.0.0.1Hi, 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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