Hello Daniel (and of course the whole community
J),
thanks for the reply. Now it works in most cases.
But I have one SIP client which handles the SIP communication very accurate and this client is actual internal registered. If I make a call from an external client to this internal client the following
error is coming.
Error message: INVITE message: Remote host ‘193.16.163.58’ is not matching with configured register server ‘203.207.111.58’ or proxy server
And no call is possible.
If I change this special client with another SIP client it works, but I can see in the debugging information that the SIP server is not routing in the desired way.
In the INVITE message the URI is not with the URI from the internal network (debugging information below) and this is the reason the special client is responding with “Error
message”. In the SDP the URI IP address is correct.
My question is: Is it possible to change the URI in the To header to the right domain?
Cheers,
Kai Ohnacker
My setting is:
Kamailio with two interfaces in two domains.
Internal: 203.207.111.58
External: 193.16.163.58
My config:
Route{
if(is_method("REGISTER")) {
if(dst_ip==203.207.111.58){
setbflag(20);
}else{
setbflag(21);
}
}
}
And
Route [NATMANAGE]
if(dst_ip==203.207.111.58 && isbflagset(20)){
rtpproxy_manage("coii");
}else if(dst_ip==203.207.111.58 && isbflagset(21)){
rtpproxy_manage("coie");
}else if(dst_ip==193.16.163.58 && isbflagset(20)){
rtpproxy_manage("corei");
}else if(dst_ip==193.16.163.58 && isbflagset(21)){
rtpproxy_manage("coee");
}
Debugging information at the destination client (internal):
INVITE sip:8000@10.99.217.42:5060 SIP/2.0
Record-Route: <sip:203.207.111.58;r2=on;lr=on;nat=yes>
Record-Route: <sip:193.16.163.58;r2=on;lr=on;nat=yes>
Via: SIP/2.0/UDP 203.207.111.58;branch=z9hG4bK9ed6.83de5667d5ceb951adb58a0f7d6909cd.0
Via: SIP/2.0/UDP 193.16.163.39:38771;rport=44934;branch=z9hG4bK802b882b077be511bce30c32c5410478
From: "PhonerLite" <sip:5000@193.16.163.58>;tag=2462361638
To: sip:8000@193.16.163.58
<- this URI IP is wrong. It should be 8000@203.207.111.58
Call-ID: 802B882B-077B-E511-BCE1-0C32C5410478@172.16.216.174
CSeq: 37 INVITE
Contact: <sip:5000@193.16.163.39:38771;alias=193.16.163.39~44934~1>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 69
Supported: 100rel, replaces, from-change
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:5000@193.16.163.58>
Content-Type: application/sdp
Content-Length: 331
v=0
o=- 618137597 1 IN IP4
203.207.111.58
s=SIPPER for PhonerLite
c=IN IP4 203.207.111.58
t=0 0
m=audio 20038 RTP/AVP 9 8 0 107 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:636280517
a=sendrecv
a=nortpproxy:yes
-------------------------------------------
11:57:25,299: T: 203.207.111.58:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.207.111.58;branch=z9hG4bK9ed6.83de5667d5ceb951adb58a0f7d6909cd.0
Via: SIP/2.0/UDP 193.16.163.39:38771;rport=44934;branch=z9hG4bK802b882b077be511bce30c32c5410478
Record-Route: <sip:203.207.111.58;r2=on;lr=on;nat=yes>
Record-Route: <sip:193.16.163.58;r2=on;lr=on;nat=yes>
From: "PhonerLite" <sip:5000@193.16.163.58>;tag=2462361638
To: <sip:8000@193.16.163.58>;tag=802b882b077be511b35b430b8303a094
Call-ID: 802B882B-077B-E511-BCE1-0C32C5410478@172.16.216.174
CSeq: 37 INVITE
Contact: <sip:8000@10.99.217.42:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Supported: replaces, from-change
Server: SIPPER for PhonerLite
Content-Length: 309
v=0
o=- 2073314527 1 IN IP4
10.99.217.42 <- client specific IP address
s=SIPPER for PhonerLite
c=IN IP4 10.99.217.42
t=0 0
m=audio 5062 RTP/AVP 8 107 0 9 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:906481595
a=sendrecv
Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Gesendet: Mittwoch, 21. Oktober 2015 12:57
An: Ohnacker, Kai [CBC] <kai.ohnacker@cbc.de>; sr-users@lists.sip-router.org
Betreff: Re: AW: [SR-Users] Routing between 2 domains
Hello,
I spotted at least one issue: you set the branch flag (with setbflag()) but then you test message/transaction flags with isflagset() -- to test branch flags you have to use isbflagset().
Cheers,
Daniel
On 21/10/15 11:42, kai.ohnacker@cbc.de wrote:
Hello Daniel,
thanks for the reply. I try out your suggestion, but this is not working. There is no sound transmitted and the connection ended after 23 seconds (I think a timer is expired).
Scenario is:
Kamailio route all internal calls only internal (external vice versa)
Kamailio route all internal to external through rtpproxy (also vice versa)
INT_IP: 203.207.111.58
EXT_IP: 193.16.163.58
Here my code:
request_route {
if(is_method("REGISTER")) {
if(dst_ip==203.207.111.58){
setbflag(20);
}else{
setbflag(21);
}
}
}
Route [NATMANAGE]
if(dst_ip==203.207.111.58 && isflagset(20)){
rtpproxy_manage("coii");
}else if(dst_ip==203.207.111.58 && isflagset(21)){
rtpproxy_manage("coie");
}else if(dst_ip==193.16.163.58 && isflagset(20)){
rtpproxy_manage("coei");
}else if(dst_ip==193.16.163.58 && isflagset(21)){
rtpproxy_manage("coee");
}
Alternative try
if(from_uri=~".*@203.207.111.58" && isflagset(20)){
rtpproxy_manage("coii");
}else if(from_uri=~".*@203.207.111.58" && isflagset(21)){
rtpproxy_manage("coie");
}else if(from_uri=~".*@193.16.163.58" && isflagset(20)){
rtpproxy_manage("coei");
}else if(from_uri=~".*@193.16.163.58" && isflagset(21)){
rtpproxy_manage("coee");
}
Another try
if(from_uri=~".*@203.207.111.58" && isflagset(20)){
rtpproxy_manage("coii");
}else if(from_uri=~".*@203.207.111.58" && isflagset(21)){
rtpproxy_manage("coie");
}else if(from_uri=~".*@193.16.163.58" && isflagset(20)){
rtpproxy_manage("coei");
}else if(from_uri=~".*@193.16.163.58" && isflagset(21)){
rtpproxy_manage("coee");
}
RTPproxy config
EXTRA_OPTS="-l 203.207.111.58/193.16.163.58 -m 20000 -M 20100 -d WARN:LOG_LOCAL1"
Some debugging information are in the txt file. TCPdump debugging does not shows interesting informations…
Has somebody a good idea which could be helpful? Do you need some more information?
Cheers,
Kai
Von: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla
Gesendet: Montag, 12. Oktober 2015 14:26
An: Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org>
Betreff: Re: [SR-Users] Routing between 2 domains
Hello,
the src_ip is not a local IP, so do not match it with INT_IP or EXT_IP. The dst_ip is local ip, but the one on which the interface was received.
You can test the $fs to see what socket is going to be used for sending out.
Or set some branch flag for each interface when the registration is processed and check that in branch route
if(is_method("REGISTER")) {
if(dst_ip==INT_IP){
setbflag(20);
}else{
setbflag(21);
}
}
then in route[NATMANGE] have conditions like:
if(src_ip==INT_IP && isflagset(20)) {
# internal to internal
} else if(src_ip==INT_IP && isflagset(21)) {
# internal to external
} else if ...
Cheers,
DanielOn 08/10/15 11:14, kai.ohnacker@cbc.de wrote:
Hello community,
I work with Kamailio 4.2.0 which is placed between two networks (with two interfaces) and RTPproxy in bridge mode.
The task from Kamailio is to handle the calls from internal networks to internal networks only on the internal Interface without bridging. For external networks vice versa. If the call comes from external to internal, then the RTPproxy should work in bridge mode and vice versa. I adapt the example with IPv4 and IPv6 routing, but this is not working for internal to external calls and vice versa (http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). Only external calls and only internal calls works. I think the cause of this behavior is the routing which I implemented to the RTPmanage part in the Kamailio config.
The error message is given by the internal device. (call from external to internal)
If I make a call from internal to external, the RTP stream is only in one direction.
Internal Interface IP: 203.207.111.58
External Interface IP: 193.16.163.58
The RTPproxy config is:
CONTROL_SOCK=udp:127.0.0.1:9000
EXTRA_OPTS="-l 203.207.111.58/193.16.163.58 -d WARN:LOG_LOCAL1"
Kamailio.cfg:
#!define INT_IP 10.96.0.0/14
#!define EXT_IP 193.0.0.0/8
# RTPProxy control and singaling updates for NAT traversal
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
# rtpproxy_manage("co");
# Start Test routing
if((src_ip==INT_IP && dst_ip==EXT_IP)){
rtpproxy_manage("cowie");
}
if((src_ip==EXT_IP && dst_ip==INT_IP)){
rtpproxy_manage("cowei");
}
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
set_contact_alias();
}
}
#!endif
return;
}
Can someone help me with the right routing? Do you need some more debugging information?
Cheers,
Kai
_______________________________________________SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/micondaBook: SIP Routing With Kamailio - http://www.asipto.com
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com