Hello Salvatore,
Looks like the server is asking for a VFU update during the call using
the INFO message. The VFU is only for video, therefore you do not get
the message when you remove the video codec from the initial setup
request and your application works in standard audio only mode.
In your uac_modified.xml, you should add support for the 'INFO' message
as an optional incoming request and respond to it accordingly (200 or
488, etc.). If you are using RTP in the application, then your h263
stream will need to set the proper marker bit to indicate an I-Frame
when you receive the VFU request. Alternatively, you can disable VFU
requests from your server if that is an option.
Regards,
Kalpan
Salvatore Frandina wrote:
---------- Forwarded message ----------
From: *Salvatore Frandina* <salvatore.frand...@gmail.com
<mailto:salvatore.frand...@gmail.com>>
Date: 2010/1/21
Subject: App_Conference: SIPp SDP and DTMF mode
To: Neil Stratford <ne...@vipadia.com <mailto:ne...@vipadia.com>>,
Mihai Balea <mi...@hates.ms <mailto:mi...@hates.ms>>,
kape...@ns1.jnetdns.de <mailto:kape...@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:[servi...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:s...@[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 '1-3...@127.0.1.1 <mailto:1-3...@127.0.1.1>':
while pausing (index 5), received 'INFO sip:s...@127.0.1.1:5061
<http://sip:s...@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:9...@192.168.0.4:5060
<http://sip:9...@192.168.0.4:5060>>;tag=as47ad7951
To: sipp <sip:s...@127.0.1.1:5061 <http://sip:s...@127.0.1.1:5061>>;tag=1
Contact: <sip:9...@192.168.0.4 <mailto:sip%3a9...@192.168.0.4>>
Call-ID: 1-3...@127.0.1.1 <mailto:1-3...@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.frand...@gmail.com <mailto:salvatore.frand...@gmail.com>
_______________________________________
--
_______________________________________
Salvatore Frandina
website: http://frandinas.altervista.org
mail: salvatore.frand...@gmail.com <mailto:salvatore.frand...@gmail.com>
_______________________________________
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------------------------------------------------------------------------------
Throughout its 18-year history, RSA Conference consistently attracts the
world's best and brightest in the field, creating opportunities for Conference
attendees to learn about information security's most important issues through
interactions with peers, luminaries and emerging and established companies.
http://p.sf.net/sfu/rsaconf-dev2dev
_______________________________________________
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