This is a known isssue - a patch was already posted in a previous mail
to this list:
http://sourceforge.net/mailarchive/forum.php?thread_name=5e64a8f90807300
925i31b3f20dje634bfe911e6f79c%40mail.gmail.com&forum_name=sipp-users
/Matt
________________________________
From: Taylor [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 23 September 2008 7:36 AM
To: [email protected]
Subject: [Sipp-users] Possible bug with SIPp pertaining to uac rtp ports
Hello all,
I'm encountering a problem when trying to run tests with Asterisk. I
have created an Asterisk config such that:
SIPp (uac_pcap) <----> Asterisk <-----> SIPp (uas)
SIP traffic is flowing perfectly, however, I can't get RTP media to
flow. The media is leaving the uac_pcap instance of SIPp and then is
rejected by the Asterisk server with an ICMP packet saying that the
destination UDP port is not open.
The RTP port, to the best of my knowledge, is negotiated (and generally
dictated by the server) at the time of call construction via SIP. At
this point, I figured either that Asterisk was sending an incorrect UDP
port for media, or SIPp was ignoring this port.
After some tcpdumps and asterisk sip debugs, it looks to be the latter.
Asterisk is sending the SIPp client a five digit UDP port number, and
SIPp is then sending media to a four digit port, which is essentially
the five digit port with the last digit (1s digit) truncated.
Here is the output I am seeing:
Asterisk output specifying the RTP port:
"""""""""""""
<------------>
Audio is at 192.168.100.25 port 14272
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx*CLI>
<--- Reliably Transmitting (NAT) to 192.168.100.156:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.100.156:5060;branch=z9hG4bK-16464-1-0;received=192.168.100.156
From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=16464SIPpTag091
To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as462c2628
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> >
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2124 2124 IN IP4 192.168.100.25
s=session
c=IN IP4 192.168.100.25
t=0 0
m=audio 14272 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
""""""""""""""""
RTP traffic leaving SIPp computer(192.168.100.156) and going to
asterisk(192.168.100.25):
"""""""""""""
14:11:03.125895 IP 192.168.100.156.6000 > 192.168.100.25.1427: UDP,
length 252
14:11:03.156076 IP 192.168.100.156.6000 > 192.168.100.25.1427: UDP,
length 252
14:11:03.186269 IP 192.168.100.156.6000 > 192.168.100.25.1427: UDP,
length 252
"""""""""""""
So 14272 was specified as the port, but SIPp is sending it to 14272.
Here are the two listings of parameters that I am using to start SIPp:
UAC:
sudo ./sipp -s s -sn uac_pcap -p 5060 -i 192.168.100.156 192.168.100.25
-r 0 -l 1 -mi 192.168.100.156
UAS:
./sipp -sn uas -p 5061 -mp 6001 -mi 192.168.100.156 -i 192.168.100.156
-rtp_echo
Is this a bug, or am I doing something wrong? Ideas?
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