[asterisk-dev] Fwd: App_Conference: SIPp SDP and DTMF mode

[Salvatore Frandina]

---------- Forwarded message ----------
From: Salvatore Frandina <salvatore.frandina at gmail.com>
Date: 2010/1/21
Subject: App_Conference: SIPp SDP and DTMF mode
To: Neil Stratford <neils at vipadia.com>, Mihai Balea <mihai at hates.ms>,
kapejod at ns1.jnetdns.de



Hi,

I'm using SIPp application http://sipp.sourceforge.net/ to generate a SIP
call to open source PBX Asterisk.

Command to call the extension (extension) where the IP is IP address of
Asterisk
[code]sipp -m 1 -d 36000000 -s extension -sf uac_modified.xml IP [/code]

In the configuration file uac_modified.xml there are the following lines

[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 at 127.0.1.1': while pausing (index 5), received
'INFO sip:sipp at 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 at 192.168.0.4:5060>;tag=as47ad7951
To: sipp <sip:sipp at 127.0.1.1:5061>;tag=1
Contact: <sip:9999 at 192.168.0.4 <sip%3A9999 at 192.168.0.4>>
Call-ID: 1-3005 at 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
[code] 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 DTMF mode works well and there is no error.

The problem it's difficult can you help me?

Thank you very much

-- 
_______________________________________
Salvatore Frandina
website: http://frandinas.altervista.org
mail: salvatore.frandina at gmail.com

_______________________________________




-- 
_______________________________________
Salvatore Frandina
website: http://frandinas.altervista.org
mail: salvatore.frandina at gmail.com

_______________________________________
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