Forgot to add,
 
you may need to rewritehost (or even contact information) to point at the Asterisk server before you do the forwarding to Asterisk.

 


On 20/12/06, Max Gregorian <gregorian442@googlemail.com> wrote:
I think I may have spotted the problem in the signalling.
 
In this reply to the INVITE, the first stage is fine:
 
Asterisk -> OpenSER (The signalling here is fine from Asterisk)
 
 
In the second stage, the reply from OpenSER to UAC

OpenSER -> Client
 
this part
 
Session Description Protocol
           Session Description Protocol Version (v): 0
           Owner/Creator, Session Id (o): root 20137 20138 IN IP4
OOO.OOO.OOO.OOO
           Session Name (s): session
           Connection Information (c): IN IP4 Session Description Protocol
           Session Description Protocol Version (v): 0
           Owner/Creator, Session Id (o): root 20137 20138 IN IP4
OOO.OOO.OOO.OOO
           Session Name (s): session
           Connection Information (c): IN IP4 OOO.OOO.OOO.OOO
           Time Description, active time (t): 0 0
           Media Description, name and address (m): audio 58508           Time Description, active time (t): 0 0
           Media Description, name and address (m): audio 58508

 
you see where it says IN IP4, this should still have the IP of Asterisk if you want the RTP to go to the Asterisk server, but instead it has the IP of OpenSER so RTP is sent to OpenSER.
 
This information is taken from somewhere (I forget - i think it's the contact of the REGISTER message), but it can be fixed.
 
You will need to look at the entire call flow starting from when the UAC first registers with OpenSER. Then, in your NAT handling routine use I think fix_nated_register and fix_nated_contact .
 
Hope this helps.
 
 
 
 
On 12/19/06, Edoardo Serra <osdevel@webrainstorm.it > wrote:
At 13.51 19/12/2006, Klaus Darilion wrote:
>You said that the 200 contains openser's IP in the SDP? Is it put in
>there by openser or already by Asterisk?

Tnx very much for help

It's put in there by OpenSER.

I'm attaching the 2 SIP/SDP packets (1 from asterisk to openser and 1
from openser to client)

AAA.AAA.AAA.AAA stands for IP of Asterisk
OOO.OOO.OOO.OOO stands for IP of OpenSER
CCC.CCC.CCC.CCC stands for IP of client
3333333333 is the called number


No.     Time        Source                Destination           Protocol Info
    20 12.646925   AAA.AAA.AAA.AAA       OOO.OOO.OOO.OOO       SIP/SDP
Status: 200 OK, with session description

Session Initiation Protocol
   Status-Line: SIP/2.0 200 OK
   Message Header
       Via: SIP/2.0/UDP
OOO.OOO.OOO.OOO;branch=z9hG4bK5bbd.eaf4f093.0;received=OOO.OOO.OOO.OOO
       Via: SIP/2.0/UDP
CCC.CCC.CCC.CCC:8952;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
       Record-Route: <sip:OOO.OOO.OOO.OOO ;lr=on;ftag=9043ec70>
       From: "test"<sip:test@OOO.OOO.OOO.OOO>;tag=9043ec70
       To: "3333333333"< sip:3333333333@OOO.OOO.OOO.OOO>;tag=as30a7528b
       Call-ID: 98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU.
       CSeq: 3 INVITE
       User-Agent: Asterisk
       Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
       Contact: <sip:3333333333@AAA.AAA.AAA.AAA:5060>
       Content-Type: application/sdp
       Content-Length: 291
   Message body
       Session Description Protocol
           Session Description Protocol Version (v): 0
           Owner/Creator, Session Id (o): root 20137 20138 IN IP4
AAA.AAA.AAA.AAA
           Session Name (s): session
           Connection Information (c): IN IP4 AAA.AAA.AAA.AAA
           Time Description, active time (t): 0 0
           Media Description, name and address (m): audio 58508
RTP/AVP 98 3 8 0 101
           Media Attribute (a): rtpmap:98 iLBC/8000
           Media Attribute (a): rtpmap:3 GSM/8000
           Media Attribute (a): rtpmap:8 PCMA/8000
           Media Attribute (a): rtpmap:0 PCMU/8000
           Media Attribute (a): rtpmap:101 telephone-event/8000
           Media Attribute (a): fmtp:101 0-16
           Media Attribute (a): silenceSupp:off - - - -

No.     Time        Source                Destination           Protocol Info
    21 12.647437   OOO.OOO.OOO.OOO       CCC.CCC.CCC.CCC       SIP/SDP
Status: 200 OK, with session description

Session Initiation Protocol
   Status-Line: SIP/2.0 200 OK
   Message Header
       Via: SIP/2.0/UDP
OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
       Record-Route: <sip:OOO.OOO.OOO.OOO;lr=on;ftag=9043ec70>
       From: "test"< sip:test@OOO.OOO.OOO.OOO>;tag=9043ec70
       To: "3333333333"<sip:3333333333@OOO.OOO.OOO.OOO >;tag=as30a7528b
       Call-ID: 98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU.
       CSeq: 3 INVITE
       User-Agent: Asterisk
       Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
       Contact: <sip:3333333333@AAA.AAA.AAA.AAA:5060>
       Content-Type: application/sdp
       Content-Length: 291
   Message body
       Session Description Protocol
           Session Description Protocol Version (v): 0
           Owner/Creator, Session Id (o): root 20137 20138 IN IP4
OOO.OOO.OOO.OOO
           Session Name (s): session
           Connection Information (c): IN IP4 OOO.OOO.OOO.OOO
           Time Description, active time (t): 0 0
           Media Description, name and address (m): audio 58508
RTP/AVP 98 3 8 0 101
           Media Attribute (a): rtpmap:98 iLBC/8000
           Media Attribute (a): rtpmap:3 GSM/8000
           Media Attribute (a): rtpmap:8 PCMA/8000
           Media Attribute (a): rtpmap:0 PCMU/8000
           Media Attribute (a): rtpmap:101 telephone-event/8000
           Media Attribute (a): fmtp:101 0-16
           Media Attribute (a): silenceSupp:off - - - -

Tnx very much for help again

Regards

Edoardo


>regards
>klaus
>
>
>regards
>klaus
>
>Edoardo Serra wrote:
>>Hi guys,
>>     I'm having a problem with an OpenSER acting as registrar
>> server and load balancer for many Asterisk servers.
>>In a few words: "users are registering on openser and, when they
>>want to make a call, OpenSER proxies the request to an Asterisk
>>server with the dispatcher module"
>>Here is the intended data flow (SIP goes through OpenSER and media
>>goes directly to Asterisk)
>>User <-- SIP --> OpenSER <-- SIP --> Asterisk
>>User <-- RTP --> Asterisk
>>Both, OpenSER and Asterisks have public IPs
>>I already have a working setup of that and everything seems working
>>correctly.
>>I'm trying to replicate that setup on another site, same
>>configurations of the boxes, same versions of OpenSER and Asterisk,
>>etc... but I'm having monodirectional Audio.
>>Having a look with tethereal I see that OpenSER, when the
>>communication is answered, sends a SIP packet (200 OK) to the user
>>indicating itself as media endpoint instead of the Asterisks.
>>  From that moment I see RTP packets flowing from the client to OpenSER
>>This seems really strange to me because I just copied the same
>>configurations file from a working setup to the new installation.
>>Tnx in advance for help.
>>Regards
>>P.S.: Here is my openser.cfg
>>## $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
>>## simple quick-start config script
>>#
>># ----------- global configuration parameters ------------------------
>>#debug=3 # debug level (cmd line: -dddddddddd)
>>fork=yes
>>#log_stderror=no # (cmd line: -E)
>>check_via=no # (cmd. line: -v)
>>dns=no # (cmd. line: -r)
>>rev_dns=no # (cmd. line: -R)
>>#children=4
>>#port=5060
>>fifo="/tmp/ser_fifo"
>>#uid=nobody
>>#gid=nobody
>># ------------------ module loading ----------------------------------
>>loadmodule "/usr/lib/openser/modules/sl.so"
>>loadmodule "/usr/lib/openser/modules/tm.so"
>>loadmodule "/usr/lib/openser/modules/rr.so"
>>loadmodule "/usr/lib/openser/modules/maxfwd.so"
>>loadmodule "/usr/lib/openser/modules/usrloc.so"
>>loadmodule "/usr/lib/openser/modules/registrar.so"
>>loadmodule "/usr/lib/openser/modules/nathelper.so"
>>loadmodule "/usr/lib/openser/modules/textops.so"
>>loadmodule "/usr/lib/openser/modules/exec.so"
>>loadmodule "/usr/lib/openser/modules/uri.so"
>>loadmodule "/usr/lib/openser/modules/uri_db.so"
>>loadmodule "/usr/lib/openser/modules/dispatcher.so"
>># Uncomment this if you want digest authentication
>># mysql.so must be loaded !
>>loadmodule "/usr/lib/openser/modules/mysql.so"
>>loadmodule "/usr/lib/openser/modules/auth.so"
>>loadmodule "/usr/lib/openser/modules/auth_db.so"
>>modparam("usrloc", "db_mode", 2)
>>modparam("usrloc", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser")
>>modparam("usrloc", "timer_interval", 120)
>>modparam("auth_db", "calculate_ha1", 0)
>>modparam("auth_db", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/voip")
>>modparam("uri_db", "db_url", " mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser")
>>modparam("rr", "enable_full_lr", 1)
>>modparam("registrar", "nat_flag", 6)
>>modparam("registrar", "max_expires", 3600)
>>modparam("registrar", "min_expires", 60)
>>modparam("registrar", "append_branches", 0)
>>modparam("registrar", "desc_time_order", 1)
>>modparam("nathelper", "natping_interval", 20) # Ping interval 20 s
>>modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
>>modparam("dispatcher", "force_dst", 1)
>># ------------------------- request routing logic -------------------
>># main routing logic
>>route{
>># initial sanity checks -- messages with
>># max_forwards==0, or excessively long requests
>>if (!mf_process_maxfwd_header("10")) {
>>sl_send_reply("483","Too Many Hops");
>>exit;
>>};
>>if ( msg:len > max_len ) {
>>sl_send_reply("513", "Message too big");
>>exit;
>>};
>>if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) {
>>sl_send_reply("405", "Method Not Allowed");
>>exit;
>>}
>>if (!method=="REGISTER") {
>>record_route();
>>};
>>if ((src_ip==xxx.xxx.xxx.xxx) || (src_ip==xxx.xxx.xxx.xxx)) { # IP
>>of Asterisks
>>if (!lookup("location")) {
>>sl_send_reply("404", "Not Found");
>>exit;
>>};
>># forward to current uri now; use stateful forwarding; that
>># works reliably even if we forward from TCP to UDP
>>if (!t_relay()) {
>>sl_reply_error();
>>};
>>exit;
>>};
>>if (nat_uac_test("3")) {
>>if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) {
>>fix_nated_contact();
>>force_rport();
>>setflag(6); # Mark as NATed
>>}
>>}
>># if the request is for other domain use UsrLoc
>># (in case, it does not work, use the following command
>># with proper names and addresses in it)
>>if (method=="REGISTER") {
>>if (!proxy_authorize("domain", "openser_view")) {
>>proxy_challenge("domain", "0");
>>exit;
>>}
>>if (!check_to()) {
>>sl_send_reply("403", "Digest username and URI username do NOT
>>match! Stay away!");
>>exit;
>>}
>>save("location");
>>exit;
>>};
>>
>>if (method=="INVITE") {
>>if (!proxy_authorize("domain", "openser_view")) {
>>proxy_challenge("domain", "0");
>>exit;
>>}
>>if (!check_from()) {
>>sl_send_reply("403", "Digest username and URI username do NOT
>>match! Stay away!");
>>exit;
>>}
>>}
>># loose-route processing
>>if (loose_route()) {
>># mark routing logic in request
>>append_hf("P-hint: rr-enforced\r\n");
>>route(1);
>>exit;
>>};
>>if (!uri==myself) {
>># mark routing logic in request
>>append_hf("P-hint: outbound\r\n");
>>route(1);
>>exit;
>>};
>>append_hf("P-hint: usrloc applied\r\n");
>>route(1);
>>}
>>route[1]
>>{
>># ! Nathelper
>>if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
>>!search("^Route:")){
>>sl_send_reply("479", "We don't forward to private IP addresses");
>>exit;
>>};
>># NAT processing of replies; apply to all transactions (for example,
>># re-INVITEs from public to private UA are hard to identify as
>># NATed at the moment of request processing); look at replies
>>t_on_reply("1");
>># send it out now; use stateful forwarding as it works reliably
>># even for UDP2TCP
>>if ((src_ip!= xxx.xxx.xxx.xxx) && (src_ip!=xxx.xxx.xxx.xxx)) { # IP
>>of Asterisks
>>ds_select_dst("2", "0");
>>}
>>if (!t_relay()) {
>>sl_reply_error();
>>};
>>}
>># ! Nathelper
>>onreply_route[1] {
>># NATed transaction ?
>>if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
>>fix_nated_contact();
>># otherwise, is it a transaction behind a NAT and we did not
>># know at time of request processing ? (RFC1918 contacts)
>>} else if (nat_uac_test("1")) {
>>fix_nated_contact();
>>};
>>}
>>
>>_______________________________________________
>>Users mailing list
>>Users@openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
>--
>Klaus Darilion
>nic.at
>


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