Hi Ian,
Please find attached the patch available at my side.
Thanks,
Arjun
----- Original Message -----
From: "Arjun Shankar K S" <[email protected]>
To: "OpenSIPS users mailling list" <[email protected]>
Cc: "OpenSIPS users mailling list" <[email protected]>
Sent: Monday, June 11, 2012 4:11:25 PM GMT +05:30 Chennai, Kolkata, Mumbai, New
Delhi
Subject: Re: [OpenSIPS-Users] No Voice Comm in Conference call
Hi Ian,
Sincere thanks for your reply.
I am attaching a patch herewith. Please let me know if you are referring to the
same and this patch did not work for me. If it works for you can you pl let me
know wat are changes that has to be made in the opensip,cfg file?
Else if patch is different, pl send me the patch.
Regards,
Arjun
----- Original Message -----
From: "Ian Buckner" <[email protected]>
To: "OpenSIPS users mailling list" <[email protected]>
Sent: Monday, June 11, 2012 3:48:40 PM GMT +05:30 Chennai, Kolkata, Mumbai, New
Delhi
Subject: Re: [OpenSIPS-Users] No Voice Comm in Conference call
Hi Arjun,
Regarding rtpproxy - if it makes your life easier, there is a very simple patch
for the rtpproxy source which allows you to specify an advertised IP address on
the command line this will be returned to OpenSips and inserted into SDP rather
than the listening address. I used this last week for a server behind NAT and
it worked perfectly.
If you want to mail me I'll happily send you a patched tarball of source.
best,
Ian
On 11 Jun 2012, at 11:14, Bogdan-Andrei Iancu wrote:
Hi Arjun,
If you have no audio at all for the call to Conf Server, you need to check the
signaling, particularity speaking the SDP part to be sure you inserted RTPProxy
correctly between UAC and Conf Server.
So, make a SIP capture for the call to Conf and check if the IPs in SDP are
correct.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com
On 06/07/2012 05:46 PM, Arjun Shankar K S wrote:
Hi All,
Greetings to everyone !!!
I have set up opensips and RTP Proxy in two different hosts since I have
opensips in a Natted environment where RTP Proxy refused to budge.
Now I have installed RTP Proxy in a direct public IP. Normal calls between 2
Client is working great !!
During conference call, the calls get connected but there is no voice
communication between any of them and soon the client who was connected last,
gets disconnected.
I could not find much support regarding this issue. Any support is sincerely
appreciated.
In my opensips.cfg, I have made the following config changes for RTP in
different host,
------Nat Params-------------
modparam("usrloc","nat_bflag", 6)
modparam("nathelper","rtpproxy_sock", "udp:rtp_proxy_publicIP:7890")
modparam("nathelper","natping_interval", 30)
modparam("nathelper","ping_nated_only", 0)
modparam("nathelper","sipping_bflag", 7)
modparam("nathelper","sipping_from", "sip:pinger@PROXY_IP" )
modparam("registrar","received_avp", "$avp(i:42)")
modparam("nathelper","received_avp", "$avp(i:42)")
I am running my RTP Proxy using the following command,
./rtpproxy -l rtp_proxy_publicIP -s udp:*:7890 -F
Thanks,
Arjun
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users diff -r rtpproxy-1.2.1/main.c rtpproxy-1.2.1-thamer/main.c
132a133
> cf->advertised = NULL;
150c151
< while ((ch = getopt(argc, argv, "vf2Rl:6:s:S:t:r:p:T:L:m:M:u:Fin:Pad:")) != -1)
---
> while ((ch = getopt(argc, argv, "vf2Rl:6:s:S:t:r:p:T:L:m:M:u:Fin:Pad:A:")) != -1)
151a153,157
>
> case 'A':
> cf->advertised = strdup(optarg);
> break;
>
diff -r rtpproxy-1.2.1/rtpp_command.c rtpproxy-1.2.1-thamer/rtpp_command.c
204c204,209
< else
---
> else {
>
> if(cf->advertised != NULL)
> len += sprintf(cp, "%d %s%s\n", lport, cf->advertised,
> (lia[0]->sa_family == AF_INET) ? "" : " 6");
> else
206a212,214
>
> }
>
diff -r rtpproxy-1.2.1/rtpp_defines.h rtpproxy-1.2.1-thamer/rtpp_defines.h
138a139,140
>
> char *advertised;
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