SRTP errors when testing using SIPp

[rw290]

Hey,
i’m trying to run a basic stressing scenario. I setup TLS using this guide.
I setup the following
extensions.conf

[from-internal]
exten = service,1,Answer(500)
same = n,Playback(demo-instruct)
same = n,Hangup()

pjsip.conf

[transport-tls]
type = transport
protocol = tls
bind = 192.168.1.10:5061
cert_file = /etc/asterisk/keys/asterisk.crt
priv_key_file = /etc/asterisk/keys/asterisk.key
method=tlsv1_2

[16001]
type=endpoint
context=from-internal
allow=all
auth=
aors=16001
media_encryption=sdes

[16001]
type=auth
auth_type=userpass
password=password
username=16001

[16001]
type=aor
max_contacts=20

i’m using this xml as input

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

<!-- UC360 INVITE/200/ACK/BYE/200 sample UAC scenario -->

<scenario name="Basic UC360 UAC">

  <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: 16001 <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
      To: <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 10 INVITE
      Contact: <sip:16001@[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Max-Forwards: 70
      User-Agent: VIRTUAL Mitel-UC-Endpoint (Mitel UC360 Collaboration Point/2.1.0.99;  08:00:0F:74:80:E1)
      Subject: Conference
      Session-Expires: 3600;refresher=uas
      Min-SE: 90
      Supported: 100rel
      Require: 100rel
      Content-Length: [len]

      v=0
      o=16001 0 0 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [rtpstream_audio_port] RTP/AVP 0 18 9 103 8 101
      a=crypto:[cryptotag1audio] [cryptosuiteaescm128sha1321audio] inline:[cryptokeyparams1audio]
      a=crypto:[cryptotag2audio] [cryptosuiteaescm128sha1802audio] inline:[cryptokeyparams2audio]
      a=rtcp:[rtpstream_audio_port+1]
      a=sendrecv
      a=rtpmap:0 PCMU/8000
      a=rtpmap:18 G729/8000
      a=rtpmap:9 G722/16000
      a=fmtp:9 bitrate=64000
      a=rtpmap:103 G7221/16000
      a=fmtp:103 bitrate=32000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16

    ]]>
  </send>

  <recv response="100"
        optional="true">
  </recv>


  <!-- receive 200 OK / PRACK (180 Ringing) -->
  <recv response="200">
  </recv>

  <!-- receive 200 OK / INVITE -->
  <recv response="200">
  </recv>

  <!-- NOTE:  [branch-5] is used to specify reuse of same [branch] value that was used five messages earlier (e.g. INVITE) -->
  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-5]
      From: "16001" <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
      To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 10 ACK
      Content-Length: 0

    ]]>
  </send>
  <nop>
      <action>
          <exec rtp_stream="apattern,1,0,PCMU/8000" />
      </action>
  </nop>

  <pause milliseconds="20000" />

  <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: "16001" <sip:16001@[remote_ip]:[remote_port]>;tag=[call_number]
      To: <sip:[service]@[remote_ip]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 12 BYE
      Contact: <sip:16001@[local_ip]:[local_port];transport=[transport]>
      Max-Forwards: 70
      Subject: Conference
      User-Agent: VIRTUAL Mitel-UC-Endpoint (Mitel UC360 Collaboration Point/2.1.0.99;  08:00:0F:74:80:E1)
      Content-Length: 0

    ]]>
  </send>

  <recv response="200">
  </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>
running command
./sipp -m 5 -sf ../sipp_scenarios/pfca_uac_apattern_crypto_simple_aescm128sha132.xml -t ln -au sipp2 -ap sipp2 -rtpcheck_debug -srtpcheck_debug -trace_err -tls_cert ../sipp.pem  -tls_key ../sipp.pem -max_socket 10  -trace_err 192.168.1.10:5061

i’m getting SRTP decrypt errors when using -m bigger then 1

  == SRTP unprotect failed on SSRC 1938815232 because of authentication failure 160
  == SRTP unprotect failed on SSRC 1762511176 because of authentication failure 160
  == SRTP unprotect failed on SSRC 1037862561 because of authentication failure 160

any idea?