I'm presently working on a SIP setup whereby there are 3 Cisco routers which each have analog phones connected to them via FXS ports. All 3 of these routers are connected via an underlying network. I have a machine hanging off from one of these routers running ser. For ease of labelling, I'll call these routers: router1, router2, and router3 (SIP server directly connected to this router via ethernet). I'm attempting to setup call forking using the UsrLoc database (this will eventually be SQL, but for the sake of the short-term I'm just storing UsrLoc in memory). The desired call forking setup looks something like this:
router1 --> router2 --> router 3
router 2 --> router 1 --> router 3
router 3 --> router 1 --> router 2
I am able to complete calls between router1 and router3 (and vice-versa) and carry on a conversation, but when calling between router1 and router2 the call completes, but neither party can hear the other. Ironically, router1 and router2 are sitting right next to each other (though, connected via another router). However, The SIP proxy is directly connected to router3. Doing a 'debug voip rtp' I see RTP messages travel bidirectionally in a constant stream with correct IP addresses and ports until the call ends, but at no point during the conversation can either party hear the other. This would lead me to believe that something other than SIP was at play, but when I bypass the proxy (point the two routers directly at each other via the dial-peer) call completion works and both parties can hear each other (I set these up as SIP, not the default H.323). Below is my ser.cfg file and the output of 'serctl ul show' for the static UsrLoc entries that I've created. The routers are setup with simple dial-peers and a sip-ua.
I've verified that there isn't any type of ACL or firewall to obstruct the conversation. Every router is able to reach each other router as well as the proxy server. I'm using private address space at present, but NAT isn't being done at any point. I've pondered trying rtp_proxy and forcing the bearer (RTP) traffic through the proxy, but that isn't a particularly good solution for my environment.
Any help would be greatly appreciated. I'm hoping that it's just a case of broken logic in my ser.cfg. Please CC: this address in your reply as I'm not currently on the mailing list.
Most of the configuration is derived from the sample configurations that I ran into.
---ser.cfg start---
# ----------- 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 #debug=7 #fork=no #log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.1.2 port=5060 mhomed=1 #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/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.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)
# ------------------------- request routing logic -------------------
# main routing logic alias="ser"
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; };
# 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=="INVITE") record_route();
# loose-route processing if (loose_route()) { t_relay(); 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; };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # 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(); };
}
---end ser.cfg---
---start static UsrLoc entries---
ser# ../../sbin/serctl ul show 222 200 OK <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 111 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:111@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 333 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231
---end static UsrLoc entries---
Thank you.
Forwarding this back out to the list as the original was bounced because I didn't subscribe to the list at that time.
---------- Forwarded message ---------- From: Psyber psyber1@gmail.com Date: Thu, 17 Feb 2005 18:41:04 -0500 Subject: SER Call Forking with Cisco Routers as SIP endpoints To: serusers@lists.iptel.org
I'm presently working on a SIP setup whereby there are 3 Cisco routers which each have analog phones connected to them via FXS ports. All 3 of these routers are connected via an underlying network. I have a machine hanging off from one of these routers running ser. For ease of labelling, I'll call these routers: router1, router2, and router3 (SIP server directly connected to this router via ethernet). I'm attempting to setup call forking using the UsrLoc database (this will eventually be SQL, but for the sake of the short-term I'm just storing UsrLoc in memory). The desired call forking setup looks something like this:
router1 --> router2 --> router 3
router 2 --> router 1 --> router 3
router 3 --> router 1 --> router 2
I am able to complete calls between router1 and router3 (and vice-versa) and carry on a conversation, but when calling between router1 and router2 the call completes, but neither party can hear the other. Ironically, router1 and router2 are sitting right next to each other (though, connected via another router). However, The SIP proxy is directly connected to router3. Doing a 'debug voip rtp' I see RTP messages travel bidirectionally in a constant stream with correct IP addresses and ports until the call ends, but at no point during the conversation can either party hear the other. This would lead me to believe that something other than SIP was at play, but when I bypass the proxy (point the two routers directly at each other via the dial-peer) call completion works and both parties can hear each other (I set these up as SIP, not the default H.323). Below is my ser.cfg file and the output of 'serctl ul show' for the static UsrLoc entries that I've created. The routers are setup with simple dial-peers and a sip-ua.
I've verified that there isn't any type of ACL or firewall to obstruct the conversation. Every router is able to reach each other router as well as the proxy server. I'm using private address space at present, but NAT isn't being done at any point. I've pondered trying rtp_proxy and forcing the bearer (RTP) traffic through the proxy, but that isn't a particularly good solution for my environment.
Any help would be greatly appreciated. I'm hoping that it's just a case of broken logic in my ser.cfg. Please CC: this address in your reply as I'm not currently on the mailing list.
Most of the configuration is derived from the sample configurations that I ran into.
---ser.cfg start---
# ----------- 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 #debug=7 #fork=no #log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.1.2 port=5060 mhomed=1 #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/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.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)
# ------------------------- request routing logic -------------------
# main routing logic alias="ser"
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; };
# 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=="INVITE") record_route();
# loose-route processing if (loose_route()) { t_relay(); 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; };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # 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(); };
}
---end ser.cfg---
---start static UsrLoc entries---
ser# ../../sbin/serctl ul show 222 200 OK <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 111 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:111@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 333 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231
---end static UsrLoc entries---
Thank you.
Anyone have any ideas on this? I'm considering setting up rtpproxy as a temporary workaround (assuming that will somehow mitigate the issue), but given my evironment it won't be a feasible production solution. I'm still pretty baffled as to why parties can talk between router1 and router3, router2 and router 3, but not between router1 and router2. Further confusing the situation is the fact that the call completes just fine when calling between router1 and router 2 and an RTP stream is established between the two routers, just no sound. router1 and router2 have a clear path to each other.
On Thu, 17 Feb 2005 18:41:04 -0500, Psyber psyber1@gmail.com wrote:
I'm presently working on a SIP setup whereby there are 3 Cisco routers which each have analog phones connected to them via FXS ports. All 3 of these routers are connected via an underlying network. I have a machine hanging off from one of these routers running ser. For ease of labelling, I'll call these routers: router1, router2, and router3 (SIP server directly connected to this router via ethernet). I'm attempting to setup call forking using the UsrLoc database (this will eventually be SQL, but for the sake of the short-term I'm just storing UsrLoc in memory). The desired call forking setup looks something like this:
router1 --> router2 --> router 3
router 2 --> router 1 --> router 3
router 3 --> router 1 --> router 2
I am able to complete calls between router1 and router3 (and vice-versa) and carry on a conversation, but when calling between router1 and router2 the call completes, but neither party can hear the other. Ironically, router1 and router2 are sitting right next to each other (though, connected via another router). However, The SIP proxy is directly connected to router3. Doing a 'debug voip rtp' I see RTP messages travel bidirectionally in a constant stream with correct IP addresses and ports until the call ends, but at no point during the conversation can either party hear the other. This would lead me to believe that something other than SIP was at play, but when I bypass the proxy (point the two routers directly at each other via the dial-peer) call completion works and both parties can hear each other (I set these up as SIP, not the default H.323). Below is my ser.cfg file and the output of 'serctl ul show' for the static UsrLoc entries that I've created. The routers are setup with simple dial-peers and a sip-ua.
I've verified that there isn't any type of ACL or firewall to obstruct the conversation. Every router is able to reach each other router as well as the proxy server. I'm using private address space at present, but NAT isn't being done at any point. I've pondered trying rtp_proxy and forcing the bearer (RTP) traffic through the proxy, but that isn't a particularly good solution for my environment.
Any help would be greatly appreciated. I'm hoping that it's just a case of broken logic in my ser.cfg. Please CC: this address in your reply as I'm not currently on the mailing list.
Most of the configuration is derived from the sample configurations that I ran into.
---ser.cfg start---
# ----------- 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 #debug=7 #fork=no #log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.1.2 port=5060 mhomed=1 #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/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.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)
# ------------------------- request routing logic -------------------
# main routing logic alias="ser"
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; }; # 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=="INVITE") record_route(); # loose-route processing if (loose_route()) { t_relay(); 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; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # 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(); };}
---end ser.cfg---
---start static UsrLoc entries---
ser# ../../sbin/serctl ul show 222 200 OK <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 111 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:111@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 333 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231
---end static UsrLoc entries---
Thank you.
I had a look at your previous post, but forgot to answer... My guess is that an ngrep or ethereal on the SDP part of the INVITEs between router1 and router will show that there is something wrong in the IP address or port number in the c= and m= lines of SDP. It's pretty logical: - The INVITE c= should contain router1's IP address, m= should contain the user agent's allocated RTP port - router2's OK should contain router2's IP address, and so on
You check on router1 and router2 which ports the user agents are listening on, look at the INVITE and the OK and see if they match. Compare that to the RTP stream you see, from where to where? If these ports do not match, find out why...
g-)
Psyber wrote:
Anyone have any ideas on this? I'm considering setting up rtpproxy as a temporary workaround (assuming that will somehow mitigate the issue), but given my evironment it won't be a feasible production solution. I'm still pretty baffled as to why parties can talk between router1 and router3, router2 and router 3, but not between router1 and router2. Further confusing the situation is the fact that the call completes just fine when calling between router1 and router 2 and an RTP stream is established between the two routers, just no sound. router1 and router2 have a clear path to each other.
On Thu, 17 Feb 2005 18:41:04 -0500, Psyber psyber1@gmail.com wrote:
I'm presently working on a SIP setup whereby there are 3 Cisco routers which each have analog phones connected to them via FXS ports. All 3 of these routers are connected via an underlying network. I have a machine hanging off from one of these routers running ser. For ease of labelling, I'll call these routers: router1, router2, and router3 (SIP server directly connected to this router via ethernet). I'm attempting to setup call forking using the UsrLoc database (this will eventually be SQL, but for the sake of the short-term I'm just storing UsrLoc in memory). The desired call forking setup looks something like this:
router1 --> router2 --> router 3
router 2 --> router 1 --> router 3
router 3 --> router 1 --> router 2
I am able to complete calls between router1 and router3 (and vice-versa) and carry on a conversation, but when calling between router1 and router2 the call completes, but neither party can hear the other. Ironically, router1 and router2 are sitting right next to each other (though, connected via another router). However, The SIP proxy is directly connected to router3. Doing a 'debug voip rtp' I see RTP messages travel bidirectionally in a constant stream with correct IP addresses and ports until the call ends, but at no point during the conversation can either party hear the other. This would lead me to believe that something other than SIP was at play, but when I bypass the proxy (point the two routers directly at each other via the dial-peer) call completion works and both parties can hear each other (I set these up as SIP, not the default H.323). Below is my ser.cfg file and the output of 'serctl ul show' for the static UsrLoc entries that I've created. The routers are setup with simple dial-peers and a sip-ua.
I've verified that there isn't any type of ACL or firewall to obstruct the conversation. Every router is able to reach each other router as well as the proxy server. I'm using private address space at present, but NAT isn't being done at any point. I've pondered trying rtp_proxy and forcing the bearer (RTP) traffic through the proxy, but that isn't a particularly good solution for my environment.
Any help would be greatly appreciated. I'm hoping that it's just a case of broken logic in my ser.cfg. Please CC: this address in your reply as I'm not currently on the mailing list.
Most of the configuration is derived from the sample configurations that I ran into.
---ser.cfg start---
# ----------- 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 #debug=7 #fork=no #log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.1.2 port=5060 mhomed=1 #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/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.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)
# ------------------------- request routing logic
# main routing logic alias="ser"
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; }; # 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=="INVITE") record_route(); # loose-route processing if (loose_route()) { t_relay(); 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; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # 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(); };}
---end ser.cfg---
---start static UsrLoc entries---
ser# ../../sbin/serctl ul show 222 200 OK <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 111 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:111@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 333 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231
---end static UsrLoc entries---
Thank you.
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
In case anyone sees something similar in the future... This turned out to be an IOS bug. Once I upgraded the routers to something in the 12.3(x)T train all was well.
On Thu, 17 Feb 2005 18:41:04 -0500, Psyber psyber1@gmail.com wrote:
I'm presently working on a SIP setup whereby there are 3 Ciscorouters which each have analog phones connected to them via FXS ports. All 3 of these routers are connected via an underlying network. I have a machine hanging off from one of these routers running ser. For ease of labelling, I'll call these routers: router1, router2, and router3 (SIP server directly connected to this router via ethernet). I'm attempting to setup call forking using the UsrLoc database (this will eventually be SQL, but for the sake of the short-term I'm just storing UsrLoc in memory). The desired call forking setup looks something like this:
router1 --> router2 --> router 3
router 2 --> router 1 --> router 3
router 3 --> router 1 --> router 2
I am able to complete calls between router1 and router3 (andvice-versa) and carry on a conversation, but when calling between router1 and router2 the call completes, but neither party can hear the other. Ironically, router1 and router2 are sitting right next to each other (though, connected via another router). However, The SIP proxy is directly connected to router3. Doing a 'debug voip rtp' I see RTP messages travel bidirectionally in a constant stream with correct IP addresses and ports until the call ends, but at no point during the conversation can either party hear the other. This would lead me to believe that something other than SIP was at play, but when I bypass the proxy (point the two routers directly at each other via the dial-peer) call completion works and both parties can hear each other (I set these up as SIP, not the default H.323). Below is my ser.cfg file and the output of 'serctl ul show' for the static UsrLoc entries that I've created. The routers are setup with simple dial-peers and a sip-ua.
I've verified that there isn't any type of ACL or firewall to obstruct the conversation. Every router is able to reach each other router as well as the proxy server. I'm using private address space at present, but NAT isn't being done at any point. I've pondered trying rtp_proxy and forcing the bearer (RTP) traffic through the proxy, but that isn't a particularly good solution for my environment.
Any help would be greatly appreciated. I'm hoping that it's just a case of broken logic in my ser.cfg. Please CC: this address in your reply as I'm not currently on the mailing list.
Most of the configuration is derived from the sample configurations that I ran into.
---ser.cfg start---
# ----------- 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 #debug=7 #fork=no #log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.1.2 port=5060 mhomed=1 #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/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.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)
# ------------------------- request routing logic -------------------
# main routing logic alias="ser"
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; }; # 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=="INVITE") record_route(); # loose-route processing if (loose_route()) { t_relay(); 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; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # 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(); };}
---end ser.cfg---
---start static UsrLoc entries---
ser# ../../sbin/serctl ul show 222 200 OK <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 111 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:111@<router3 IP>:5060>;q=1.00;expires=1003718231
ser# ../../sbin/serctl ul show 333 200 OK <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231 <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231
---end static UsrLoc entries---
Thank you.
-
Greger V. Teigre -
Psyber