Hi Bogdan , I have my server sip integrated with asterisk for voicemail and
meetme, adds to rtpproxy to solve my problems of nat with my remote
clients, the result was almost satisfactory some details to improve,
the problem was that after adding rtpproxy when my clients do not
answer a call no longer here jumps me to the voicemail only shows in
the 488 Not Aceptable here..<br><br>UAC == NAT == internet ===wan === eth0 server sip/asterisk === eth1 === UAC clients<br><br>asterisk port : 5070<br>sip proxy: 5060<br><br>I have tried to put the force_rtp_proxy ("", <a href="http://192.168.1.64">192.168.1.64</a>); , but it doesn't always work me on he writes the sdp twice<br>
<br>onreply_route[1] {<br><br>if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])"){<br> force_rtp_proxy();<br># force_rtp_proxy("","<a href="http://192.168.1.64">192.168.1.64</a>");<br>
append_hf("P-hint: onreply_route|force_rtp_proxy \r\n");<br><br> }<br> if (!search("^Content-Length:[ ]*0")) {<br><br># search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');<br>
<br> if (isbflagset(6)) {<br> append_hf("P-hint: Onreply-route - fixcontact \r\n");<br> fix_nated_contact();<br> }<br> }<br> exit;<br>}<br><br><br>my sip log:<br><br><pre>U +1.050293 <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> -> <a href="http://192.168.10.1:5070">192.168.10.1:5070</a><br>
INVITE <a href="http://sip:u116@192.168.10.1:5070">sip:u116@192.168.10.1:5070</a> SIP/2.0 <br>Record-Route: <sip:<a href="http://192.168.10.1">192.168.10.1</a>;lr=on;ftag=cfc7ce84fac4cb57> <br>Via: SIP/2.0/UDP <a href="http://192.168.10.1">192.168.10.1</a>;branch=z9hG4bKb305.df34945.1 <br>
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=cfc7ce84fac4cb57 <br>To: <<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>> <br>
Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes> <br>Supported: replaces, timer, path <br>Call-ID: <a href="mailto:cac52e44f0f25722@192.168.10.28">cac52e44f0f25722@192.168.10.28</a> <br>CSeq: 3157 INVITE <br>User-Agent: Grandstream GXV3000 <a href="http://1.1.3.14">1.1.3.14</a> <br>
Max-Forwards: 69 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK <br>Content-Type: application/sdp <br>Content-Length: 604 <br>P-hint: inbound->inbound <br>P-hint: Route[20]: Rtpproxy <br>
P-hint: Route[20]: Rtpproxy <br> <br>v=0 <br>o=119 8000 8001 IN IP4 <a href="http://192.168.10.28">192.168.10.28</a> <br>s=SIP Call <br><b>c=IN IP4 192.168.10.1192.168.10.1 </b><br>t=0 0 <br>m=audio 3505635056 RTP/AVP 18 4 3 2 0 101 <br>
a=sendrecv <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:4 G723/8000 <br>a=rtpmap:3 GSM/8000 <br>a=rtpmap:2 G726-32/8000 <br>a=rtpmap:0 PCMU/8000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11 <br>m=video 3505835058 RTP/AVP 99 34 <br>
a=sendrecv <br>a=rtpmap:99 H264/90000 <br>a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM48gM== <br>a=rtpmap:34 H263/90000 <br>a=fmtp:34 CIF=2 MaxBR=1280 <br>a=framerate:20 <br>
a=nortpproxy:yes <br>a=nortpproxy:yes <br><br>#<br>U +0.000051 <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> -> <a href="http://192.168.10.19:5063">192.168.10.19:5063</a><br>CANCEL <a href="http://sip:116@192.168.10.19:5063">sip:116@192.168.10.19:5063</a> SIP/2.0 <br>
Via: SIP/2.0/UDP <a href="http://192.168.10.1">192.168.10.1</a>;branch=z9hG4bKb305.df34945.0 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=cfc7ce84fac4cb57 <br>Call-ID: <a href="mailto:cac52e44f0f25722@192.168.10.28">cac52e44f0f25722@192.168.10.28</a> <br>
To: <<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>> <br>CSeq: 3157 CANCEL <br>Max-Forwards: 70 <br>Content-Length: 0 <br> <br><br>#<br>U +0.000519 <a href="http://192.168.10.1:5070">192.168.10.1:5070</a> -> <a href="http://192.168.10.1:5060">192.168.10.1:5060</a><br>
SIP/2.0 488 Not acceptable here <br>Via: SIP/2.0/UDP <a href="http://192.168.10.1">192.168.10.1</a>;branch=z9hG4bKb305.df34945.1;received=<a href="http://192.168.10.1">192.168.10.1</a> <br>Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002 <br>
From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=cfc7ce84fac4cb57 <br>To: <<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>>;tag=as3a67d233 <br>Call-ID: <a href="mailto:cac52e44f0f25722@192.168.10.28">cac52e44f0f25722@192.168.10.28</a> <br>
CSeq: 3157 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br> <br><br>#<br>U +0.000142 <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> -> <a href="http://192.168.10.1:5070">192.168.10.1:5070</a><br>
ACK <a href="http://sip:u116@192.168.10.1:5070">sip:u116@192.168.10.1:5070</a> SIP/2.0 <br>Via: SIP/2.0/UDP <a href="http://192.168.10.1">192.168.10.1</a>;branch=z9hG4bKb305.df34945.1 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=cfc7ce84fac4cb57 <br>
Call-ID: <a href="mailto:cac52e44f0f25722@192.168.10.28">cac52e44f0f25722@192.168.10.28</a> <br>To: <<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>>;tag=as3a67d233 <br>CSeq: 3157 ACK <br>Max-Forwards: 70 <br>
Content-Length: 0 <br> <br><br>#<br>U +0.000174 <a href="http://192.168.10.1:5060">192.168.10.1:5060</a> -> <a href="http://192.168.10.28:5060">192.168.10.28:5060</a><br>SIP/2.0 488 Not acceptable here <br>Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002 <br>
From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=cfc7ce84fac4cb57 <br>To: <<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>>;tag=as3a67d233 <br>Call-ID: <a href="mailto:cac52e44f0f25722@192.168.10.28">cac52e44f0f25722@192.168.10.28</a> <br>
CSeq: 3157 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br> <br><br>#<br>U +0.002025 <a href="http://192.168.10.28:5060">192.168.10.28:5060</a> -> <a href="http://192.168.10.1:5060">192.168.10.1:5060</a><br>
ACK <a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a> SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKc80584c1e2a47002 <br>From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=cfc7ce84fac4cb57 <br>
To: <<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>>;tag=as3a67d233 <br>Contact: <<a href="http://sip:119@192.168.10.28:5060">sip:119@192.168.10.28:5060</a>> <br>Proxy-Authorization: Digest username="119", realm="<a href="http://192.168.10.1">192.168.10.1</a>", algorithm=MD5, uri="<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>", nonce="491d0af68c7aade3e86ae38c262008b5141d5769", response="804c1e89ed73b32be25f010495524aca" <br>
Call-ID: <a href="mailto:cac52e44f0f25722@192.168.10.28">cac52e44f0f25722@192.168.10.28</a> <br>CSeq: 3157 ACK <br>User-Agent: Grandstream GXV3000 <a href="http://1.1.3.14">1.1.3.14</a> <br>Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK <br>
Content-Length: 0 <br> <br><br>#<br>U +0.006401 <a href="http://192.168.10.19:5063">192.168.10.19:5063</a> -> <a href="http://192.168.10.1:5060">192.168.10.1:5060</a><br>SIP/2.0 487 Request Terminated <br>To: <<a href="mailto:sip%3A116@192.168.10.1">sip:116@192.168.10.1</a>>;tag=9c88b01629d86766i3 <br>
From: <<a href="mailto:sip%3A119@192.168.10.1">sip:119@192.168.10.1</a>>;tag=cfc7ce84fac4cb57 <br>Call-ID: <a href="mailto:cac52e44f0f25722@192.168.10.28">cac52e44f0f25722@192.168.10.28</a> <br>CSeq: 3157 INVITE <br>
Via: SIP/2.0/UDP <a href="http://192.168.10.1">192.168.10.1</a>;branch=z9hG4bKb305.df34945.0 <br>Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002 <br>Record-Route: <sip:<a href="http://192.168.10.1">192.168.10.1</a>;lr=on;ftag=cfc7ce84fac4cb57> <br>
Server: Linksys/SPA942-6.1.3(a) <br>Content-Length: 0 <br></pre><br>Regards <br><br>rickygm<br><br><br><div class="gmail_quote">2008/11/13 Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">See the help from rtpproxy:<br>
<br>
usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles]<br>
<br>
and the manual for nathelper (flags for force_rtp_proxy):<br>
<a href="http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id2515879" target="_blank">http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id2515879</a><br>
<br>
</blockquote></div><br>