Hi, I am trying to build the following script:
1. Sending Invite without SDP
2. Waiting to receive 100
3. Waiting to receive 180
4. Waiting to recive 200 +SDP
The problem is that SIPP is sending ICMP port unreachable after getting 200+
SDP (see attached sniffer file)
This is the XML which are using:
<send retrans="500">
- <![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From:[field0]<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact:sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<recv response="100" optional="true" />
<recv response="180" />
<recv response="200" crlf="true" timeout="10000" />
- <!-- By adding rrs="true" (Record Route Sets), the route sets
-->
- <!-- are saved and used for following messages sent. Useful to test
-->
- <!-- against stateful SIP proxies/B2BUAs.
-->
- <!-- Packet lost can be simulated in any send/recv message by
-->
- <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.
-->
- <send>
- <![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From:[field0]<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact:sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: [len]
v=0
o=SIPp 187223291 6 IN IP4 [local_ip]
s=-
c=IN IP4 [local_ip]
b=AS:1920
t=0 0
m=audio [media_port] RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
]]>
Best Regards,
Michael Azer
[cid:image002.jpg@01D03024.DA5B3230]
+972.3.767.9372 office | +972.549701544 mobile |
www.AVAYA.com<http://www.AVAYA.com>
AO2_F1.pcap
Description: AO2_F1.pcap
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