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" <[email protected]> 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 �Csd 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 �CI lo udp �Cn �Cxxx
>
> 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:[email protected]]
> 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: [email protected]
> To: [email protected]
> 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 �Csn uas” on server 192.168.1.236, and
>
> I ran “./sipp �Csn 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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