hello friends,

sorry to disturb you people again and again iam newbie

i know ser from last two weeks only

as iam suffering with this problem from last week

i need help of you people i hope some body is kind

enough to help me out .

and my first scenario in simple ascii diagram

public ip (estara softphone) /|\ | | |/ SER server (public ip) /|\ | |/ public ip (estara softphone)

my SER server is redhat linux 9.0

iam using stable version which i got through cvs

and first checking with the public ip to publics ip

here i could able to establish the call i.e in either

side i could able to listen the voice

in this iam not using any rtpproxy ------------------------------------------------------

[root@server sbin]# ser Listening on 127.0.0.1 [127.0.0.1]:5060 <public ip>[public ip]:5060 Aliases: server.pol.net.in:5060 localhost:5060 localhost.localdomain:5060 stateless - initializing [root@server sbin]# Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 9(0) INFO: fifo process starting: 17451 9(17451) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 8(17450) parse_nameaddr(): No < found 8(17450) parse_rr(): Error while parsing name-addr 8(17450) find_first_route(): Error while parsing Route HF 6(17448) parse_nameaddr(): No < found 6(17448) parse_rr(): Error while parsing name-addr 6(17448) find_first_route(): Error while parsing Route HF 6(17448) parse_nameaddr(): No < found 6(17448) parse_rr(): Error while parsing name-addr 6(17448) find_first_route(): Error while parsing Route HF 7(17449) parse_nameaddr(): No < found 7(17449) parse_rr(): Error while parsing name-addr 7(17449) find_first_route(): Error while parsing Route HF 6(17448) parse_nameaddr(): No < found 6(17448) parse_rr(): Error while parsing name-addr 6(17448) find_first_route(): Error while parsing

------------------------------------------------------- this is the second scenario

public ip (msn messenger) /|\ | | |/ SER server (public ip) /|\ | |/ DHCP server (public ip , NAT device) /|\ | |/ private ip (estara softphone)

in the terminal of ser iam gettign this -----------------------------------------------------

[root@server sbin]# ser Listening on 127.0.0.1 [127.0.0.1]:5060 <public ip>[public ip]:5060 Aliases: server.pol.net.in:5060 localhost:5060 localhost.localdomain:5060 [root@server sbin]# stateless - initializing Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 9(0) INFO: fifo process starting: 17642 9(17642) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 5(17638) parse_nameaddr(): No < found 5(17638) parse_rr(): Error while parsing name-addr 5(17638) find_first_route(): Error while parsing Route HF 8(17641) parse_nameaddr(): No < found 8(17641) parse_rr(): Error while parsing name-addr 8(17641) find_first_route(): Error while parsing Route HF 5(17638) ERROR: extract_body: message body has lenght zero 5(17638) ERROR: force_rtp_proxy: can't extract body from the message 5(17638) ERROR: on_reply processing failed

----------------------------------------------------- here iam using the rtpproxy of version 1.4 2003/08/05

./rtpproxy -f

in the terminal of rtpproxy iam getting this ----------------------------------------------------- [root@server rtpproxy]# ./rtpproxy -f rtpproxy: new session on a port 35000 rtpproxy: lookup on a port 35000 rtpproxy: addr1 filled in: 202.65.128.24 rtpproxy: addr2 filled in: 202.65.148.252 rtpproxy: stats: 179 in from addr1, 3 in from addr2, 180 relayed rtpproxy: session on port 35000 is cleaned up

------------------------------------------------------

the result is i could able to see that both mic and

speaker are working and iam listening what ever public

ip softphone is speaking in private ip softphone.

but in public ip softphone i have seen that only

mic is working not the speakers i.e i could not

able to listen what ever private ip softphone is

speaking

i observed that one malformed sip packet is genrating

thorugh the ser in tethereal

so my ethereal report is

**************************************************** Frame 3 (327 bytes on wire, 327 bytes captured) Arrival Time: May 23, 2004 20:18:03.259293000 Time delta from previous packet: 2.401569000 seconds Time relative to first packet: 2.402778000 seconds Frame Number: 3 Packet Length: 327 bytes Capture Length: 327 bytes Ethernet II, Src: 00:e0:2b:90:1f:00, Dst: 00:e0:18:ed:04:61 Destination: 00:e0:18:ed:04:61 (Asustek__ed:04:61) Source: 00:e0:2b:90:1f:00 (Extreme__90:1f:00) Type: IP (0x0800) Internet Protocol, Src Addr: 202.*.*.252 (dhcp i.e nat server) (202.*.*.252), Dst Addr: 202. *.*.19 (202.*.*.19) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 0000 00.. = Differentiated Services Codepoint: Default (0x00) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 313 Identification: 0xf5f8 Flags: 0x00 .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 125 Protocol: UDP (0x11) Header checksum: 0x9d28 (correct) Source: 202.*.*.252 (202.*.*.252) Destination: 202.*.*.19 (202.*.*.19) User Datagram Protocol, Src Port: 62812 (62812), Dst Port: 5060 (5060) Source port: 62812 (62812) Destination port: 5060 (5060) Length: 293 Checksum: 0x8112 (correct) Session Initiation Protocol Request line: REGISTER sip:202.*.*.19 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.18:5060 From: sip:203@202.*.*.19 To: sip:203@202.*.*.19 Contact: sip:203@192.168.1.18:5060 Call-ID: e166bf60-acf2-11d8-aafe-00e01846e257@192.168.1.18 CSeq: 32320301 REGISTER Content-Length: 0 Expires: 3600

Frame 4 (629 bytes on wire, 629 bytes captured) Arrival Time: May 23, 2004 20:18:03.261431000 Time delta from previous packet: 0.002138000 seconds Time relative to first packet: 2.404916000 seconds Frame Number: 4 Packet Length: 629 bytes Capture Length: 629 bytes Ethernet II, Src: 00:e0:18:ed:04:61, Dst: 00:e0:2b:90:1f:00 Destination: 00:e0:2b:90:1f:00 (Extreme__90:1f:00) Source: 00:e0:18:ed:04:61 (Asustek__ed:04:61) Type: IP (0x0800) Internet Protocol, Src Addr: 202.*.*.19 (202.*.*.19), Dst Addr: 202.*.*.252 (202.*.*.252) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 615 Identification: 0x0000 Flags: 0x04 .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x8ee3 (correct) Source: 202.*.*.19 (202.*.*.19) Destination: 202.*.*.252 (202.*.*.252) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62812 (62812) Source port: 5060 (5060) Destination port: 62812 (62812) Length: 595 Checksum: 0x0d6c (correct) Session Initiation Protocol Status line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.18:5060;rport=62812;received=202.*.*.252 From: sip:203@202.*.*.19 To: sip:203@202.65.128.19;tag=b27e1a1d33761e85846fc98f5f3a7e58.d036 Call-ID: e166bf60-acf2-11d8-aafe-00e01846e257@192.168.1.18 CSeq: 32320301 REGISTER Status line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.18:5060;rport=62812;received=202.*.*.252 From: sip:203@202.*.*.19 To: sip:203@202.*.*.19;tag=b27e1a1d33761e85846fc98f5f3a7e58.d036 Call-ID: e166bf60-acf2-11d8-aafe-00e01846e257@192.168.1.18 CSeq: 32320301 REGISTER Contact: sip:203@202.*.*.252:62812;q=0.00;expires=3600 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 202.*.*.19:5060 "Noisy feedback tells: pid=17639 req_src_ip=202.*.*.252 req_src_p ort=62812 in_uri=sip:202.*.*.19 out_uri=sip:202.65.128.19 via_cnt==1"

Frame 5 (46 bytes on wire, 46 bytes captured) Arrival Time: May 23, 2004 20:18:18.476274000 Time delta from previous packet: 15.214843000 seconds Time relative to first packet: 17.619759000 seconds Frame Number: 5 Packet Length: 46 bytes Capture Length: 46 bytes Ethernet II, Src: 00:e0:18:ed:04:61, Dst: 00:e0:2b:90:1f:00 Destination: 00:e0:2b:90:1f:00 (Extreme__90:1f:00) Source: 00:e0:18:ed:04:61 (Asustek__ed:04:61) Type: IP (0x0800) Internet Protocol, Src Addr: 202.*.*.19 (202.*.*.19), Dst Addr: 202.*.*.252 (202.*.*.252) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 32 Identification: 0x0000 Flags: 0x04 .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x912a (correct) Source: 202.*.*.19 (202.*.*.19) Destination: 202.*.*.252 (202.*.*.252) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62812 (62812) Source port: 5060 (5060) Destination port: 62812 (62812) Length: 12 Checksum: 0x4d22 (correct) [Malformed Packet: SIP]

********************************************************* i kept * inorder to hide the public ip please pardon me for that

ser.cfg is of klaus which he kept in the we last

febaraury with some light modifications confined to my

problem

my ser.cfg is ****************************************************** # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script #

# ----------- global configuration parameters ------------------------

debug=3 # 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) port=5060 children=4 fifo="/tmp/ser_fifo" alias=server.pol.net.in #------------------ 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" loadmodule "/usr/local/lib/ser/modules/textops.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"

# load the voicemail module #loadmodule "/usr/local/lib/ser/modules/vm.so"

# load the enum module #loadmodule "/usr/local/lib/ser/modules/enum.so"

# load the group module, to verify if a user forwards to voicemail #loadmodule "/usr/local/lib/ser/modules/group.so"

# load the nathelper module loadmodule "/usr/local/lib/ser/modules/nathelper.so"

# ----------------- setting module-specific parameters ---------------

# -- registrar parameter # special NAT flag indicates that a registered client is behind NAT modparam("registrar", "nat_flag", 6)

# -- usrloc params --

#modparam("usrloc", "db_mode", 0)

# Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #/ #odparam("usrloc", "db_mode", 0) #modparam("usrloc", "db_url", "mysql://ser:XXXXXXX@localhost/ser")

# -- 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") #modparam("auth_db", "db_url", "mysql://ser:XXXXXXX@localhost/ser")

# -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)

# -- voicemail params -- #modparam("voicemail", "db_url","mysql://ser:XXXXXXX@localhost/ser")

# -- voicemail params -- #modparam("group", "db_url","mysql://serro:XXXXXXXXX@localhost/ser")

# -- nathelper params -- modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 30 ) #/modparam("register","nat_flag",6) #modparam("tm", "fr_inv_timer", 8 )

# ------------------------- request routing logic -------------------

# main routing logic

if (nat_uac_test("2")) { force_rport(); fix_nated_contact(); if (method=="INVITE") { fix_nated_sdp("1"); } append_hf("P-hint: fixed NAT contact for request\r\n"); setflag(5); }; 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; }; if (loose_route()) { log(1, "loose_route processing\n"); t_relay(); break; }; if (uri==myself) {

if (method=="REGISTER") { if (isflagset(5)) { setflag(6);

...skipping 29 lines if (method=="INVITE") { force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; }; if (!t_relay()) { sl_reply_error(); };

}

onreply_route[1] { if (isflagset(6)) { fix_nated_contact(); append_hf("P-hint: fixed NAT contact for response\r\n"); }

if ( (status=~"200" || status=~"183") ) { if ( isflagset(7) ) { force_rtp_proxy(); append_hf("P-hint: response forced to rtp proxy\r\n"); }; }; }

route[4]{ # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "REFER" || method == " BYE")) { log(1, "no invite,ack,cancel,refer->return 404\n"); sl_send_reply("404", "Not Found"); break; };

# not voicemail subscriber and no echo/conference call if ( isflagset(4)) { log(1, "flag(4) active\n"); }; if (uri =~ "conference") { log(1, "conference call\n"); }; if (uri =~ "echo") { log(1, "echo call\n"); }; if ( !( isflagset(4) || (uri =~ "conference") || (uri =~ "echo") ) ) { log(1, "no voicemail subscriber->return 404"); sl_send_reply("404", "Not Found and no voicemail turned on"); break; };

if ( isflagset(5) ) { log(1, "caller is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy"); force_rtp_proxy(); }; };

}

******************************************************

my requirement is private to public call establishment

so please kindly guide me

with regards rama kanth

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