Hello list,
I have been trying to get SIPp to send rtp audio and have run into a
roadblock. I am getting the audio on the remote end, but there is a serious
hash/buzz over the audio. You can just barely hear the proper audio sample
underneath all of the buzz. Has anyone run into this problem before?
I am including as much info as I can in hopes the answer will be as simple
as "your header is malformed" or "you recorded the audio wrongly"
I am sending the following sip header when I send the call:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.65.1.5:5061;branch=z9hG4bK-26016-1-0
From: sipp <sip:[email protected]:5061>;tag=1
To: sut <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 179
v=0
o=- 53655765 2353687637 IN IP4 10.65.1.5
s=SIPp
c=IN IP4 10.65.1.5
t=0 0
m=audio 5606 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
The server that it is connecting to (Asterisk) replies that yes, PCMU is
fine and sets the codec to g711u as I was expecting.
Previously, I had captured the audio of an entire quick call with this
command:
tcpdump -T rtp -vvv -i eth1 src 10.65.240.94 and not port 5060 -w g711u.pcap
The g711u.pcap file is what I am transmitting through the xml.
I was calling sipp via the following command line:
./sipp -sf uac_pcap.xml -d 10000 10.65.1.5 -s 555 -l 1 -m 1 -mi 10.65.1.5
-mp 5606 -i 10.65.1.5 -trace_err
E-mail me off-list if you would like the g711u.pcap file or the xml layout I
am utilizing for testing.
Thanks in advance,
Pete
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