SIPp: SDP and App_conference


I’m using SIPp application to generate a SIP call to open source PBX Asterisk. The application needs of xml configuration file where are specified configuration parameters for SIP channel. In particular i have to set the parameters for SDP protocol which manages the audio/video stream in SIP/RTP call.

Command to call the extension (extension) where the IP is IP address of Asterisk

sipp -m 1 -d 36000000 -s extension -sf uac_modified.xml IP configuration file uac_modified.xml

[code] 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]
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Dummy User
User-Agent: sipp
Content-Type: application/sdp
Content-Length: [len]

  o=sipp 53655765 2353687637 IN IP[local_ip_type] [local_ip]
  c=IN IP[local_ip_type] [local_ip]
  t=0 0
  m=audio [auto_media_port] RTP/AVP 0 97 8 18 3 101  
  a=fmtp:18 annexb=yes
  a=fmtp:101 0-11,16 
  a=rtpmap:0 PCMU/8000
  a=rtpmap:97 SPEEX/8000
  a=rtpmap:8 PCMA/8000
  a=rtpmap:18 G729/8000
  a=rtpmap:101 telephone-event/8000 
  m=video [media_port] RTP/AVP 115
  a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520
  a=rtpmap:115 H263-1998/90000

The call work well between SIPp and softphone (Eyebeam, X-lite), i can see all the messages in the Asterisk CLI.

If i try to use App_conference application the SIPp user work only without video support.
Scenarios: there is a conference (DTMF mode enabled) where there are two or more users when i press a digit to see a generic user, the SIPp user returns the following error

[code]sipp: The following events occured:
2010-01-21 16:07:10:392 1264086430.392045: Aborting call on unexpected message for Call-Id ‘1-3005@’: while pausing (index 5), received 'INFO sip:sipp@ SIP/2.0

Via: SIP/2.0/UDP;branch=z9hG4bK11e48a8e;rport

Max-Forwards: 70

From: sut sip:9999@;tag=as47ad7951

To: sipp sip:sipp@;tag=1

Contact: sip:9999@

Call-ID: 1-3005@

CSeq: 102 INFO

User-Agent: Asterisk PBX

Content-Type: application/media_control+xml

Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>










If i disable the video support without the following lines

m=video [media_port] RTP/AVP 115 a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520 a=rtpmap:115 H263-1998/90000 a=sendrecv
the DTMF mode works well and there is no error

The problem it’s difficult can you help me?