[Serusers] Re: RTPProxy fails only for Private to Public communication

7 Apr 2005

Look at this:
U 84.203.148.146:5060 -> 157.190.74.151:5060

  SIP/2.0 200 OK..Via: SIP/2.0/UDP 157.190.74.151;rport=5060;branch=z9hG4bKcd

  17ddd1b59ead49..From: "2092"
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c

  ..To: <sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID: 8ffc2d18b218

  70b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100 1.0.

  5.18..Contact: <sip:2093@172.16.3.31>..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,

  REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported: rep

  laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP

  Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP 0..a=sendrecv..

  a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..

I assume this is where you get an error message. You haven't called
fix_nated_contact() for this message, and in fact I believe there may be an error in the
ONsip.org example where a line has been lost.

192. onreply_route[1] {

193.

194. if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {

195. if (!search("^Content-Length:\ 0")) {

196. force_rtp_proxy();

197. };

198. } else if (nat_uac_test("1")) {

199. fix_nated_contact();

200. };

201. }

fix_nated_contact() should go in between line 194 and 195.  

Could you please confirm that this works?  I will look at the config file.

g-(

---- Original Message ----
From: Vivienne Curran
To: serusers(a)lists.iptel.org ; greger(a)teigre.com
Sent: Wednesday, April 06, 2005 04:12 PM
Subject: Re: RTPProxy fails only for Private to Public communication

...
  Just as an extra : I have a sniff of the message for
when a public
 client (2092)rings a private client (2093)included at the bottom of
 this email. I cant see anything wrong with them but maybe it will
 shed more light on the matter.   

 Vivienne Curran &lt;vivcurran(a)yahoo.co.uk&gt; wrote:
 I changed the line modparam("nathelper", "rtpproxy_sock",
 "/var/run/rtpproxy.sock") to modparam("nathelper",
"rtpproxy_sock",
 "udp:localhost:22222") and started the rtpproxy as ./rtpproxy -s udp
 from the relevant directory and this resulted in a series of
 "rtpp_command: no response from rtpproxy" and rtpproxy temporarily
 disabled" errors. If I return to the original modparam and start it
 as ./rtpproxy then it works but like I said when the private client
 rings the public client, I get "ERROR: send_rtpp_command: cant read
 reply from a RTP Proxy".        

 Any further ideas? Has anyone on the mailing list experienced this? I
 am using the script given in the onsip getting started doc for 0.9.0.
 but am using ser 0.8.14.  

 BR,
 Vivienne

 "Greger V. Teigre" &lt;greger(a)teigre.com&gt; wrote:
 See inline.

  Thank you for that Greger. I have altered my
script so that it
 exactly mimics the one in the onsip document besides the has_totag
 and fix_nated register. All is good when I ring from a private phone
 to a public phone i.e. the audio is very clear and the following
 messages are in /var/log.

 ERROR: extract_body: message body has length zero
 ERROR: force_rtp_proxy2: cant extract body from the message.

 I assume this is because of the 200 OK to a register message where
 theres no sdp?? Is this correct?  
 That's correct.  You will find code in the example configs where we
 test for an empty body before calling force_rtp_proxy. 

  However when I try to phone from public into
private I get:

 ERROR: send_rtpp_command: cant read reply from a RTP Proxy.

 I find this confusing because I know the rtpproxy is working.  
 This means that rtpproxy is not responding to a particular message. I
 have heard some people have had problems with the socket based
 communication. I only use UDP. This is what you do to set up udp
 (22222 is default port):   
 modparam("nathelper", "rtpproxy_sock",
"udp:localhost:22222")
 rtpproxy must be started with -s udp:*
 g-)

  BR
 Vivienne.

 "Greger V. Teigre" &lt;greger(a)teigre.com&gt; wrote:
 Yes, you can use fix_nated_contact instead. It is not entirely
 RFC-compliant, but that's what you have in 0.8.14.
 The has_totag() only tests to see if the INVITE has a To header,
 which means that it is in-dialog and thus is a re-INVITE.  An INVITE
 will normally not have loose routing unless you have another SIP
 proxy forwarding an INVITE to you (in which case you should assume
 that the other proxy handles NAT and thus not trigger NAT-related
 code).  You can safely remove the has_totag() if you use
 force_rtp_proxy("l")
 g-)

 ---- Original Message ----
 From: Vivienne Curran
 To: Greger V. Teigre ; serusers(a)lists.iptel.org
 Sent: Tuesday, April 05, 2005 02:25 PM
 Subject: Re: [Serusers] Contact Header and SDP not rewritten

> Greger,
> 
> Since fix_nated_register does not exist with 0.8.14, will
> fix_nated_contact do instead? Also if I am leaving out the
> has_totag() at the start of the script, will this greatly effect its
> functionality?
> 
> Thank you,
> Vivienne  Send instant messages to your online friends
 http://uk.messenger.yahoo.com 
 U 157.190.74.151:5060 -> 84.203.148.146:5060
   INVITE sip:2093@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
 157.190.74.151;bra 
   nch=z9hG4bKcd17ddd1b59ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aed 
   c22bd5a3b510c..To: <sip:2093@84.203.148.146>..Contact:
 <sip:2092@157.190.74 
   .151>..Supported: replaces..Call-ID:
 8ffc2d18b21870b3@157.190.74.151..CSeq: 
    64735 INVITE..User-Agent: Grandstream BT100
 1.0.5.18..Max-Forwards: 70..Al 
   low:
 INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Typ
   e: application/sdp..Content-Length: 426....v=0..o=2092 8000 0 IN
 IP4 157.19 
   0.74.151..s=SIP Call..c=IN IP4 157.190.74.151..t=0 0..m=audio 5004
 RTP/AVP 
   0 8 4 18 2 15 99 9 101..a=sendrecv..a=rtpmap:0
 PCMU/8000/3..a=rtpmap:8 PCMA 
   /8000/3..a=rtpmap:4 G723/8000/3..a=rtpmap:18
 G729/8000/3..a=rtpmap:2 G726-3 
   2/8000/3..a=rtpmap:15 G728/8000/3..a=rtpmap:99
 iLBC/8000/3..a=fmtp:99 mode= 
   20..a=rtpmap:9 G722/8000/3..a=ptime:20..a=rtpmap:101
 telephone-event/8000/3 
   ..a=fmtp:101 0-11..

 U 84.203.148.146:5060 -> 157.190.74.151:5060
   SIP/2.0 100 trying -- your call is important to us..Via:
 SIP/2.0/UDP 157.19 
   0.74.151;branch=z9hG4bKcd17ddd1b59ead49;rport=5060..From: "2092"
 <sip:2092@ 
   84.203.148.146>;tag=aedc22bd5a3b510c..To:
 <sip:2093@84.203.148.146>..Call-I 
   D: 8ffc2d18b21870b3@157.190.74.151..CSeq: 64735 INVITE..Server: Sip
 EXpress 
    router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
 84.203.148.1 
   46:5060 "Noisy feedback tells:  pid=8990 req_src_ip=157.190.74.151
 req_src_ 
   port=5060 in_uri=sip:2093@84.203.148.146
 out_uri=sip:2093@84.203.148.14:506 
   0 via_cnt==1"....

 U 84.203.148.146:5060 -> 84.203.148.14:5060
   INVITE sip:2093@84.203.148.14:5060 SIP/2.0..Via: SIP/2.0/UDP
 84.203.148.146 
   ;branch=z9hG4bKf51e.b169be72.0..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060; 
   branch=z9hG4bKcd17ddd1b59ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag= 
   aedc22bd5a3b510c..To: <sip:2093@84.203.148.146>..Contact:
 <sip:2092@157.190 
   .74.151:5060>..Supported: replaces..Call-ID:
 8ffc2d18b21870b3(a)157.190.74.15 
   1..CSeq: 64735 INVITE..User-Agent: Grandstream BT100
 1.0.5.18..Max-Forwards 
   : 69..Allow:
 INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Con 
   tent-Type: application/sdp..Content-Length: 445....v=0..o=2092 8000
 0 IN IP 
   4 157.190.74.151..s=SIP Call..c=IN IP4 84.203.148.146..t=0
 0..m=audio 35014 
    RTP/AVP 0 8 4 18 2 15 99 9 101..a=sendrecv..a=rtpmap:0
 PCMU/8000/3..a=rtpm 
   ap:8 PCMA/8000/3..a=rtpmap:4 G723/8000/3..a=rtpmap:18
 G729/8000/3..a=rtpmap 
   :2 G726-32/8000/3..a=rtpmap:15 G728/8000/3..a=rtpmap:99
 iLBC/8000/3..a=fmtp 
   :99 mode=20..a=rtpmap:9 G722/8000/3..a=ptime:20..a=rtpmap:101
 telephone-eve 
   nt/8000/3..a=fmtp:101 0-11..a=nortpproxy:yes..

 U 84.203.148.14:5060 -> 84.203.148.146:5060
   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
 84.203.148.146;branch=z9hG4bKf51e.b169 
   be72.0..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b5 
   9ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <s 
   ip:2093@84.203.148.146>..Call-ID:
 8ffc2d18b21870b3@157.190.74.151..CSeq: 64 
   735 INVITE..User-Agent: Grandstream BT100 1.0.5.18..Content-Length:
 0.... 

 U 84.203.148.14:5060 -> 84.203.148.146:5060
   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
 84.203.148.146;branch=z9hG4bKf51e.b16 
   9be72.0..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b 
   59ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: < 
   sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d18b21870b3@15 
   7.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100
 1.0.5.18..C 
   ontent-Length: 0....

 U 84.203.148.146:5060 -> 157.190.74.151:5060
   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG 
   4bKcd17ddd1b59ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3 
   b510c..To: <sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d1 
   8b21870b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent:
 Grandstream BT100 
    1.0.5.18..Content-Length: 0....

 U 84.203.148.14:5060 -> 84.203.148.146:5060
   SIP/2.0 200 OK..Via: SIP/2.0/UDP
 84.203.148.146;branch=z9hG4bKf51e.b169be72 
   .0..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b59ead 
   49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <sip:2 
   093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d18b21870b3(a)157.190 
   .74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100
 1.0.5.18..Contac 
   t: <sip:2093@172.16.3.31>..Allow:
 INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION 
   S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported:
 replaces..Conte 
   nt-Length: 152....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=SIP
 Call..c=IN I 
   P4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP
 0..a=sendrecv..a=rtpmap:0 PCMU/ 
   8000/3..a=ptime:20..

 U 84.203.148.146:5060 -> 157.190.74.151:5060
   SIP/2.0 200 OK..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd 
   17ddd1b59ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c 
   ..To: <sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d18b218 
   70b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream
 BT100 1.0. 
   5.18..Contact: <sip:2093@172.16.3.31>..Allow:
 INVITE,ACK,CANCEL,BYE,NOTIFY, 
   REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
 application/sdp..Supported: rep 
   laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4
 172.16.3.31..s=SIP 
   Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP
 0..a=sendrecv.. 
   a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..

 U 84.203.148.14:5060 -> 84.203.148.146:5060
   SIP/2.0 200 OK..Via: SIP/2.0/UDP
 84.203.148.146;branch=z9hG4bKf51e.b169be72 
   .0..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b59ead 
   49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <sip:2 
   093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d18b21870b3(a)157.190 
   .74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100
 1.0.5.18..Contac 
   t: <sip:2093@172.16.3.31>..Allow:
 INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION 
   S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported:
 replaces..Conte 
   nt-Length: 152....v=0..o=2093 8000 1 IN IP4 172.16.3.31..s=SIP
 Call..c=IN I 
   P4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP
 0..a=sendrecv..a=rtpmap:0 PCMU/ 
   8000/3..a=ptime:20..

 U 84.203.148.146:5060 -> 157.190.74.151:5060
   SIP/2.0 200 OK..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd 
   17ddd1b59ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c 
   ..To: <sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d18b218 
   70b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream
 BT100 1.0. 
   5.18..Contact: <sip:2093@172.16.3.31>..Allow:
 INVITE,ACK,CANCEL,BYE,NOTIFY, 
   REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
 application/sdp..Supported: rep 
   laces..Content-Length: 174....v=0..o=2093 8000 1 IN IP4
 172.16.3.31..s=SIP 
   Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP
 0..a=sendrecv.. 
   a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..

 U 84.203.148.14:5060 -> 84.203.148.146:5060
   SIP/2.0 200 OK..Via: SIP/2.0/UDP
 84.203.148.146;branch=z9hG4bKf51e.b169be72 
   .0..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b59ead 
   49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <sip:2 
   093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d18b21870b3(a)157.190 
   .74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100
 1.0.5.18..Contac 
   t: <sip:2093@172.16.3.31>..Allow:
 INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION 
   S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported:
 replaces..Conte 
   nt-Length: 152....v=0..o=2093 8000 2 IN IP4 172.16.3.31..s=SIP
 Call..c=IN I 
   P4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP
 0..a=sendrecv..a=rtpmap:0 PCMU/ 
   8000/3..a=ptime:20..

 U 84.203.148.146:5060 -> 157.190.74.151:5060
   SIP/2.0 200 OK..Via: SIP/2.0/UDP
 157.190.74.151;rport=5060;branch=z9hG4bKcd 
   17ddd1b59ead49..From: "2092"
 <sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c 
   ..To: <sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
 8ffc2d18b218 
   70b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream
 BT100 1.0. 
   5.18..Contact: <sip:2093@172.16.3.31>..Allow:
 INVITE,ACK,CANCEL,BYE,NOTIFY, 
   REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
 application/sdp..Supported: rep 
   laces..Content-Length: 174....v=0..o=2093 8000 2 IN IP4
 172.16.3.31..s=SIP 
   Call..c=IN IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP
 0..a=sendrecv.. 
   a=rtpmap:0 PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..

 Send instant messages to your online friends
 http://uk.messenger.yahoo.com  

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[Serusers] Re: RTPProxy fails only for Private to Public communication