Wrong source port used by sipp to send media when using rtp_stream

[atsakiridis]

Even though SDP negotiation is successful and sipp declares port 6000 for media in the SDP offer, for some reason when sending RTP packets it doesn't use as source port 6000, while it does receive them in that port when in the receiving end. That causes a problem to the other end that expects media to be transferred using the port negotiated in SDP.

Using https://github.com/SIPp/sipp/releases/tag/v3.5.0 on OSX 10.11.2

Sorry, forgot to add some more data:

• sipp command used: sipp -sf ./health-ivr-sipp-client.xml -s +5555 127.0.0.1:5080 -mi 127.0.0.1:5090 -l 1 -m 1 -r 1 -trace_screen -trace_err -recv_timeout 5000 -nr -t u1
• sipp xml file:

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="UAC with media"> 
<!-- Declare Variables -->
<Global variables="clientId,clientIdString" /> 
<!-- Initialize Variables -->
<init>
   <nop>
      <action>
         <assign assign_to="clientId" value="0" />
      </action>
   </nop>
</init> 
<!-- Increment Counter and Convert to String -->
<nop>
   <action>
      <add assign_to="clientId" value="1" />
      <assignstr assign_to="clientIdString" value="[$clientId]" />
      <ereg regexp="[0-9]+" search_in="var" variable="clientIdString" assign_to="clientIdString" check_it="true" />
   </action>
</nop>

<!-- send INVITE -->
<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=[call_number]
To: sut <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
X-ClientId: [$clientIdString]
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 [media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
]]>
</send>

<!-- receive TRYING -->
<recv response="100" optional="true" />

<!-- receive RINGING -->
<recv response="180" optional="true" />

<!-- receive OK -->
<!-- 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" />

<!-- send ACK -->
<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=[call_number]
To: sut <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>

<!-- Wait 5 seconds for the announcements to play -->
<pause milliseconds="5000"/>

<!-- Wait 2 seconds to simulate ringing -->
<pause milliseconds="2000"/>

<!-- Simulate sending RTP traffic for one minute -->
<nop>
<action>
<!--
<exec rtp_stream="../../resources/audio/demo-prompt-one-minute.wav"/>
-->
<exec play_pcap_audio="../../resources/pcap/message.pcap"/>
</action>
</nop>

<pause milliseconds="60000"/>

<!-- Send BYE to terminate the call -->
<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=[call_number]
To: sut  <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">
</recv>

<!-- STATISTICS -->
<!-- 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>

[wdoekes]

Thank you for taking the time to report. We'll look into it.

[atsakiridis]

One important note I forgot to mention is that with play_pcap_audio it works fine, which is why my .xml above is a bit misleading as I copy/pasted after we found this workaround

Please let me know if you need more info

Best regards,
Antonis

[wdoekes]

@atsakiridis: Your best bet is probably to use [rtpstream_audio_port] in place of [media_port] in your SDP. That should work.

[atsakiridis]

Thanks a lot @wdoekes, I'll check it out

[stadelmannj]

Hello, using SIPp 3.5.2, I still see the same issue. thoutht this problem is solved with latest 3.5.2 ?

[norouzzadegan]

Hi i have problem of rtcp port
my pcap file sample is sample.pcap that include rtp and rtcp packet
I want send rtcp packet on other port that equal media_port+1 in a scenario but i cant send rtcp packet on other port.