Hi there,
I have the following:
SER:
version: ser 0.9.0 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE
262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c
compiled on 16:46:51 Mar 29 2005 with gcc 2.96
CISCO GATEWAY:
IOS (tm) 5300 Software (C5300-IS-M), Version 12.2(26), RELEASE SOFTWARE
(fc2)
The situation is the following:
If I have to UAs, both behind same (linux-iptables) nat, I have a
two-way audio communication, with nice quality, with no problems at all.
If I place a call to the PSTN, trough my Gateway, from my UA, I have a
two-way audio communication, with nice quality, with no problems at all.
If I try to receive a call from the PSTN, trough my Gateway, I only have
a one-way audio communication, but only in the SIP->PSTN direction. What
could this be?.
I can send my ser.cfg and some dumps from communications, but I believe
the problem is quite clear. In fact, I followed the instructions from
onsip.org.
- some ngrep's ...
U ip_ua:5070 -> ip_ser_mediaproxy:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP ip_ser_mediaproxy;branch=z9hG4
U ip_ser_mediaproxy:5060 -> snom_proxy:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP snom_proxy:5060;branch=z9
U snom_proxy:5060 -> ip_ser_mediaproxy:5060
ACK sip:ip_ser_mediaproxy:5060 SIP/2.0..v: SIP/2.0/UDP 38.117.2
U ip_ser_mediaproxy:5060 -> ip_ua:5070
ACK sip:1991015@ip_ua:5070 SIP/2.0..Record-Route: <
The ACK arrives nicely to my UA (ip_ua:5070), but it keeps sending RTP
to my GATEWAY's IP and port.
- tcpdumping RTP on the UA (private IP:192.168.1.185.15090)(behind NAT)
20:36:37.596851 192.168.1.185.15090 > IP_CISCO.20482: udp 32
- show call active voice brief at my cisco:
2D2B : 104269100hs.1 +273 pid:1991334 Originate 1991015 active dur
00:00:36 tx:162/3202 rx:1790/35800 IP ip_ser_mediaproxy:35234 rtt:0ms
pl:33170/140ms lost:0/0/0 delay:69/69/70ms g729r8
- session.py (from media proxy) (look at the ports, its fine and logic
according to the tcpdumping of RTP at my LAN).
Caller Via Called Status Duration
Codec Type Traffic
------------------------------------------------------------------------
---------------------------
IP_CISCO:20482 - ip_ser_mediaproxy:35234 - ?.?.?.?:? inactive 1'04"
G729 Audio 11.29k/0/0
I would appreciate some help ( or even IDEAS )
Best regards,
Lucas
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