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<DIV><FONT face=Arial size=2><SPAN class=515132810-23102007>Hello
all,</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=515132810-23102007></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=515132810-23102007>I am noticing that
OpenSER relays an INVITE and the packet that is sent out is clipped. You can see
in the following packets that the forwarded packet is missing a few bytes from
the SDP part. Counting the bytes shows that the outgoing packet is always 1512
bytes. Is this something that has do with OpenSER (module parameter, compile
options) or is it OS related?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=515132810-23102007></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=515132810-23102007>thank you for any
help</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=515132810-23102007></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=515132810-23102007>George</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>U 2007/10/23 13:27:46.165232 213.5.1.6:57665 ->
213.5.43.4:5060<BR>INVITE sip:1012118204501@213.5.43.4:5060 SIP/2.0.<BR>Via:
SIP/2.0/UDP
213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C1AB9.<BR>Remote-Party-ID:
"GEOP Papadopoul"
<sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off.<BR>From:
"GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821.<BR>To:
<sip:1012118204501@213.5.43.4>.<BR>Date: Tue, 23 Oct 2007 10:27:46
GMT.<BR>Call-ID: <A
href="mailto:6E2DF805-808911DC-AA258232-F1669B65@213.5.1.6">6E2DF805-808911DC-AA258232-F1669B65@213.5.1.6</A>.<BR>Supported:
100rel,timer,resource-priority,replaces.<BR>Min-SE: 1800.<BR>Cisco-Guid:
1848010793-2156466652-3218142496-4228086245.<BR>User-Agent:
Cisco-SIPGateway/IOS-12.x.<BR>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER.<BR>CSeq: 101
INVITE.<BR>Max-Forwards: 70.<BR>Timestamp: 1193135266.<BR>Contact:
<sip:2116872933@213.5.1.6:5060>.<BR>Call-Info:
<sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000".<BR>Expires:
180.<BR>Allow-Events: telephone-event.<BR>Content-Type:
application/sdp.<BR>Content-Disposition:
session;handling=required.<BR>Content-Length:
290.<BR>.<BR>v=0.<BR>o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4
213.5.1.6.<BR>s=SIP Call.<BR>c=IN IP4 213.5.1.6.<BR>t=0 0.<BR>m=audio 16454
RTP/AVP 18 0 8 100.<BR>c=IN IP4 213.5.1.6.<BR>a=rtpmap:18
G729/8000.<BR>a=fmtp:18 annexb=yes.<BR>a=rtpmap:0 PCMU/8000.<BR>a=rtpmap:8
PCMA/8000.<BR>a=rtpmap:100 X-NSE/8000.<BR>a=fmtp:100 192-194.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV><FONT face=Arial size=2>
<DIV><BR>U 2007/10/23 13:27:46.173826 213.5.43.4:5060 ->
213.5.1.6:57665<BR>SIP/2.0 100 Giving a try.<BR>Via: SIP/2.0/UDP
213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C1AB9;rport=57665.<BR>From:
"GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821.<BR>To:
<sip:1012118204501@213.5.43.4>.<BR>Call-ID: <A
href="mailto:6E2DF805-808911DC-AA258232-F1669B65@213.5.1.6">6E2DF805-808911DC-AA258232-F1669B65@213.5.1.6</A>.<BR>CSeq:
101 INVITE.<BR>Server: Altec Telecoms SIP Proxy.<BR>Content-Length:
0.<BR>.</DIV>
<DIV> </DIV>
<DIV><BR>U 2007/10/23 13:27:46.174241 213.5.43.4:5060 ->
213.5.17.226:5060<BR>INVITE sip:demo1@192.168.1.39:5060
SIP/2.0.<BR>Record-Route:
<sip:213.5.43.4;lr=on;ftag=237E8130-1821>.<BR>Via: SIP/2.0/UDP
213.5.43.4;branch=z9hG4bK4acd.5092baa7.0.<BR>Via: SIP/2.0/UDP
213.5.1.6:5060;rport=57665;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C1AB9.<BR>Remote-Party-ID:
"GEOP Papadopoul"
<sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off.<BR>From:
"GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821.<BR>To:
<sip:1012118204501@213.5.43.4>.<BR>Date: Tue, 23 Oct 2007 10:27:46
GMT.<BR>Call-ID: <A
href="mailto:6E2DF805-808911DC-AA258232-F1669B65@213.5.1.6">6E2DF805-808911DC-AA258232-F1669B65@213.5.1.6</A>.<BR>Supported:
100rel,timer,resource-priority,replaces.<BR>Min-SE: 1800.<BR>Cisco-Guid:
1848010793-2156466652-3218142496-4228086245.<BR>User-Agent:
Cisco-SIPGateway/IOS-12.x.<BR>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER.<BR>CSeq: 101
INVITE.<BR>Max-Forwards: 10.<BR>Timestamp: 1193135266.<BR>Contact:
<sip:2116872933@213.5.1.6:57665>.<BR>Call-Info:
<sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000".<BR>Expires:
180.<BR>Allow-Events: telephone-event.<BR>Content-Type:
application/sdp.<BR>Content-Disposition:
session;handling=required.<BR>Content-Length: 290.<BR>P-hint: NATed client
request.<BR>.<BR>v=0.<BR>o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4
213.5.1.6.<BR>s=SIP Call.<BR>c=IN IP4 213.5.1.6.<BR>t=0 0.<BR>m=audio 16454
RTP/AVP 18 0 8 100.<BR>c=IN IP4 213.5.1.6.<BR>a=rtpmap:18
G729/8000.<BR>a=fmtp:18 annexb=yes.<BR>a=rtpmap:0 PCMU/8000.<BR>a=rtpmap:8
PCMA/8000.<BR>a=rtpmap:100 X-NSE/8000.<BR>a=fmtp:100
<BR></FONT></DIV></BODY><!--[object_id=#altectelecoms.gr#]--><FONT face=Tahoma size=2><FONT color=#0000ff>
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