Hi,
I’m using SIPp application sipp.sourceforge.net/ 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]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Dummy User
User-Agent: sipp
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sipp 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 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
a=sendrecv
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
[/code]
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@127.0.1.1’: while pausing (index 5), received 'INFO sip:sipp@127.0.1.1:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK11e48a8e;rport
Max-Forwards: 70
From: sut sip:9999@192.168.0.4:5060;tag=as47ad7951
To: sipp sip:sipp@127.0.1.1:5061;tag=1
Contact: sip:9999@192.168.0.4
Call-ID: 1-3005@127.0.1.1
CSeq: 102 INFO
User-Agent: Asterisk PBX 1.6.2.0
Content-Type: application/media_control+xml
Content-Length: 205
<?xml version="1.0" encoding="utf-8" ?><media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>
'.[/code]
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?