Hi,
After some testing on the latest release, i have some problem doing the following on LINUX :
Scenario : - SIP Phones behind a NAT - SER server under linux with rtpproxy launched - a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere.
I could not find a sample ser.cfg script that reflect this scenario. Could someone send me this scenario ?
Maybe i missunderstood some things. In particular, do i need to launch two instances of ser (one for outbound proxy, another for request. If yes, how to do that)
Thanks.
olivier@siteboulevard.com wrote:
Hi,
After some testing on the latest release, i have some problem doing the following on LINUX :
latest? du you mean unstable or latest stable?
Scenario :
- SIP Phones behind a NAT
- SER server under linux with rtpproxy launched
- a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere.
Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port of the rtpproxy. If this is correct, you should see RTP streams to rtpproxy (which should be forwarded to the GW and the NAT box)
I could not find a sample ser.cfg script that reflect this scenario. Could someone send me this scenario ?
this is like any other scenario with a client behind NAT and one client with public IP.
Maybe i missunderstood some things. In particular, do i need to launch two instances of ser (one for outbound proxy, another for request. If yes, how to do that)
You don't need two instances.
Klaus
OK, thanks.
On the following conf (in fact, the NAT example), where should i put the rewritehost and forward function to my CISCO ??
# # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support
# This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so"
# !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2)
# -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password")
# -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
# !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# ------------------------- 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"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; };
# !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received #if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER
if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n");
# This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling".
fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; #};
# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; };
if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; };
# 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 (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # };
save("location"); break; };
lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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"); break; };
# if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); };
# 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 (!t_relay()) { sl_reply_error(); }; }
# !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else { fix_nated_contact(); }; }
olivier@siteboulevard.com wrote:
Hi,
After some testing on the latest release, i have some problem doing the following on LINUX :
latest? du you mean unstable or latest stable?
Scenario :
- SIP Phones behind a NAT
- SER server under linux with rtpproxy launched
- a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere.
Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port of the rtpproxy. If this is correct, you should see RTP streams to rtpproxy (which should be forwarded to the GW and the NAT box)
I could not find a sample ser.cfg script that reflect this scenario. Could
someone send me this scenario ?
this is like any other scenario with a client behind NAT and one client with public IP.
Maybe i missunderstood some things. In particular, do i need to launch two
instances of ser (one for outbound proxy, another for request. If yes, how
to
do that)
You don't need two instances.
Klaus
olivier@siteboulevard.com wrote:
OK, thanks.
On the following conf (in fact, the NAT example), where should i put the rewritehost and forward function to my CISCO ??
In front of the lookup("alias") I would check if the username is numerical, then I would format it (according to the local dial plan) to an E.164 number. After that I would do an ENUM lookup. If after the ENUM lookup the request-URI is still an E.164 number, I would rewrite the host.
Otherwise do the lookup-alias and lookup location.
In but cases, the message will be forwarded by the t_relay at the end of your script.
Klaus
# # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support
# This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so"
# !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2)
# -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password")
# -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
# !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# ------------------------- 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"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received #if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP,rewriting\n");
# This will work only for user agents that support symmetric # communication. We tested quite many of them and majorityis # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling".
fix_nated_contact(); # Rewrite contact with source IP ofsignalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; #};
# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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 (uri==myself) { if (method=="REGISTER") {# Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # };
save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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"); break; };
# if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # 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 (!t_relay()) { sl_reply_error(); };}
# !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else { fix_nated_contact(); }; }
olivier@siteboulevard.com wrote:
Hi,
After some testing on the latest release, i have some problem doing the following on LINUX :
latest? du you mean unstable or latest stable?
Scenario :
- SIP Phones behind a NAT
- SER server under linux with rtpproxy launched
- a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere.
Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port of the rtpproxy. If this is correct, you should see RTP streams to rtpproxy (which should be forwarded to the GW and the NAT box)
I could not find a sample ser.cfg script that reflect this scenario. Could
someone send me this scenario ?
this is like any other scenario with a client behind NAT and one client with public IP.
Maybe i missunderstood some things. In particular, do i need to launch two
instances of ser (one for outbound proxy, another for request. If yes, how
to
do that)
You don't need two instances.
Klaus
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Selon Klaus Darilion klaus.mailinglists@pernau.at:
many thanks.
In fact, everything work except no sound !
I put the debug on the rtpproxy and i see each time i call :
rtpproxy: command syntax error
I download the last src archive (not from cvs), and the last cvs version of rtpproxy.
What could it be ???
olivier@siteboulevard.com wrote:
OK, thanks.
On the following conf (in fact, the NAT example), where should i put the rewritehost and forward function to my CISCO ??
In front of the lookup("alias") I would check if the username is numerical, then I would format it (according to the local dial plan) to an E.164 number. After that I would do an ENUM lookup. If after the ENUM lookup the request-URI is still an E.164 number, I would rewrite the host.
Otherwise do the lookup-alias and lookup location.
In but cases, the message will be forwarded by the t_relay at the end of your script.
Klaus
# # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support
# This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so"
# !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2)
# -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter) # #modparam("auth_db", "password_column", "password")
# -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
# !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind
NAT
# ------------------------- 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"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received #if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP,rewriting\n");
# This will work only for user agents that supportsymmetric
# communication. We tested quite many of them andmajority
is # smart enough to be symmetric. In some phones it takes
a
configuration # option. With Cisco 7960, it is called NAT_Enable=Yes,
with kphone it is # called "symmetric media" and "symmetric signalling".
fix_nated_contact(); # Rewrite contact with source IPof
signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; #};
# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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 (uri==myself) { if (method=="REGISTER") {# Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # };
save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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");
break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (forexample,
# 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 (!t_relay()) { sl_reply_error(); };}
# !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else { fix_nated_contact(); }; }
olivier@siteboulevard.com wrote:
Hi,
After some testing on the latest release, i have some problem doing the following on LINUX :
latest? du you mean unstable or latest stable?
Scenario :
- SIP Phones behind a NAT
- SER server under linux with rtpproxy launched
- a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere.
Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port of the rtpproxy. If this is correct, you should see RTP streams to rtpproxy (which should be forwarded to the GW and the NAT box)
I could not find a sample ser.cfg script that reflect this scenario.
Could
someone send me this scenario ?
this is like any other scenario with a client behind NAT and one client with public IP.
Maybe i missunderstood some things. In particular, do i need to launch
two
instances of ser (one for outbound proxy, another for request. If yes,
how
to
do that)
You don't need two instances.
Klaus
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
If you don't need the features of the unstable version try the stable versions, check them out from cvs. For ser user rel_0_8_12 and for rtpproxy use v20040105. I use them and they are working fine!
# cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser login # cvs -z3 -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co -r rel_0_8_12 sip_router # cvs -z3 -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co -r v20040105 rtpproxy
Make sure that you don't use an older/newer binary which is hidden somewhere in your path.
Klaus
olivier@siteboulevard.com wrote:
Selon Klaus Darilion klaus.mailinglists@pernau.at:
many thanks.
In fact, everything work except no sound !
I put the debug on the rtpproxy and i see each time i call :
rtpproxy: command syntax error
I download the last src archive (not from cvs), and the last cvs version of rtpproxy.
What could it be ???
olivier@siteboulevard.com wrote:
OK, thanks.
On the following conf (in fact, the NAT example), where should i put the rewritehost and forward function to my CISCO ??
In front of the lookup("alias") I would check if the username is numerical, then I would format it (according to the local dial plan) to an E.164 number. After that I would do an ENUM lookup. If after the ENUM lookup the request-URI is still an E.164 number, I would rewrite the host.
Otherwise do the lookup-alias and lookup location.
In but cases, the message will be forwarded by the t_relay at the end of your script.
Klaus
# # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support
# This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so"
# !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2)
# -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter) # #modparam("auth_db", "password_column", "password")
# -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
# !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind
NAT
# ------------------------- 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"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received #if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP,rewriting\n");
# This will work only for user agents that supportsymmetric
# communication. We tested quite many of them andmajority
is # smart enough to be symmetric. In some phones it takes
a
configuration # option. With Cisco 7960, it is called NAT_Enable=Yes,
with kphone it is # called "symmetric media" and "symmetric signalling".
fix_nated_contact(); # Rewrite contact with source IPof
signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; #};
# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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 (uri==myself) { if (method=="REGISTER") {# Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # };
save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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");
break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (forexample,
# 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 (!t_relay()) { sl_reply_error(); };}
# !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else { fix_nated_contact(); }; }
olivier@siteboulevard.com wrote:
Hi,
After some testing on the latest release, i have some problem doing the following on LINUX :
latest? du you mean unstable or latest stable?
Scenario :
- SIP Phones behind a NAT
- SER server under linux with rtpproxy launched
- a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere.
Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port of the rtpproxy. If this is correct, you should see RTP streams to rtpproxy (which should be forwarded to the GW and the NAT box)
I could not find a sample ser.cfg script that reflect this scenario.
Could
someone send me this scenario ?
this is like any other scenario with a client behind NAT and one client with public IP.
Maybe i missunderstood some things. In particular, do i need to launch
two
instances of ser (one for outbound proxy, another for request. If yes,
how
to
do that)
You don't need two instances.
Klaus
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Selon Klaus Darilion klaus.mailinglists@pernau.at:
yes :) It works ! I took the last last last stable version of all.
Sound is really great SIP<->PSTN via a cisco.
Thanks Klaus.
If you don't need the features of the unstable version try the stable versions, check them out from cvs. For ser user rel_0_8_12 and for rtpproxy use v20040105. I use them and they are working fine!
# cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser login # cvs -z3 -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co -r rel_0_8_12 sip_router # cvs -z3 -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co -r v20040105 rtpproxy
Make sure that you don't use an older/newer binary which is hidden somewhere in your path.
Klaus
olivier@siteboulevard.com wrote:
Selon Klaus Darilion klaus.mailinglists@pernau.at:
many thanks.
In fact, everything work except no sound !
I put the debug on the rtpproxy and i see each time i call :
rtpproxy: command syntax error
I download the last src archive (not from cvs), and the last cvs version of
rtpproxy.
What could it be ???
olivier@siteboulevard.com wrote:
OK, thanks.
On the following conf (in fact, the NAT example), where should i put the rewritehost and forward function to my CISCO ??
In front of the lookup("alias") I would check if the username is numerical, then I would format it (according to the local dial plan) to an E.164 number. After that I would do an ENUM lookup. If after the ENUM lookup the request-URI is still an E.164 number, I would rewrite the host.
Otherwise do the lookup-alias and lookup location.
In but cases, the message will be forwarded by the t_relay at the end of your script.
Klaus
# # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support
# This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so"
# !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2)
# -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter) # #modparam("auth_db", "password_column", "password")
# -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
# !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind
NAT
# ------------------------- 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"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received #if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP,rewriting\n");
# This will work only for user agents that supportsymmetric
# communication. We tested quite many of them andmajority
is # smart enough to be symmetric. In some phones it
takes
a
configuration # option. With Cisco 7960, it is called
NAT_Enable=Yes,
with kphone it is # called "symmetric media" and "symmetric
signalling".
fix_nated_contact(); # Rewrite contact with source IPof
signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; #};
# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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 (uri==myself) { if (method=="REGISTER") {# Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # };
save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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");
break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (forexample,
# 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 (!t_relay()) { sl_reply_error(); };}
# !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else { fix_nated_contact(); }; }
olivier@siteboulevard.com wrote:
Hi,
After some testing on the latest release, i have some problem doing the
following on LINUX :
latest? du you mean unstable or latest stable?
Scenario :
- SIP Phones behind a NAT
- SER server under linux with rtpproxy launched
- a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound
anywhere.
Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port
of the rtpproxy. If this is correct, you should see RTP streams to rtpproxy (which should be forwarded to the GW and the NAT box)
I could not find a sample ser.cfg script that reflect this scenario.
Could
someone send me this scenario ?
this is like any other scenario with a client behind NAT and one client with public IP.
Maybe i missunderstood some things. In particular, do i need to launch
two
instances of ser (one for outbound proxy, another for request. If yes,
how
to
do that)
You don't need two instances.
Klaus
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Klaus Darilion -
olivier＠siteboulevard.com