Hi,
Can you send me your XML, because I think that the problem is not related to
ACK.
The Wireshark showing that after getting 200 message the SIPp is replying
immediately with ICMP port unreachable.
But, you are right, I missed this line and added it .
After adding this line, it is still not working.
Thanks,
Michael
From: Alberto Valente [mailto:alberto-r-vale...@telecom.pt]
Sent: Tuesday, May 05, 2015 4:15 PM
To: Azer, Michael (Michael) **CTR**; sipp-users@lists.sourceforge.net
Subject: RE: Ivite without SDP
Hi;
I have a scenario similar to yours that Works. The only difference I can see is
that you're missing the Content-Type Header in the ACK reply (
Content-Type: application/sdp)
"My" ACK is as follows:
ACK sip:[field4]@[field2] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]@[field2]>;tag=[call_number]
To: <sip:[field4]@[field2]>[peer_tag_param]
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
[last_Call-ID]
Cseq: [cseq] ACK
Content-Type: application/sdp
Content-Length: [len]
v=0
o=CiscoSystemsSIP-GW-UserAgent 2342 5515 IN IP[local_ip_type] [local_ip]
s=SIP Call
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 18 101
c=IN IP[media_ip_type] [media_ip]
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
BR
[PT]<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.telecom.pt_&d=AwMFAw&c=BFpWQw8bsuKpl1SgiZH64Q&r=Qlh7iU-afcg2NgiTv4ti9w&m=izY26WJsAszxpKV2gFCXN3SsvkECmZjxZT9Xa3k__Rc&s=CwaO14wXoX2KVsc4IalWhZ21O3XzvQQqUlrBSkGXHLg&e=>
Alberto Manuel Rolim Valente
alberto.r.vale...@telecom.pt<mailto:alberto.r.vale...@telecom.pt>
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From: Azer, Michael (Michael) [mailto:ma...@avaya.com]
Sent: terça-feira, 5 de Maio de 2015 11:30
To: sipp-users@lists.sourceforge.net<mailto:sipp-users@lists.sourceforge.net>
Subject: [Sipp-users] Ivite without SDP
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 |
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