Re: [Freeswitch-users] rtp timeout yet
to be sure I'm silly I downgraded from trunk 17001 and rtp timeout disappeared. Regards ----- Original Message ----- From: Madovsky To: freeswitch-users@lists.freeswitch.org Sent: Monday, March 22, 2010 3:20 PM Subject: Re: [Freeswitch-users] rtp timeout yet so when I use a public IP to register a phone, there s a rtp timeout from line 853 of switch_rtp.c I looked at the code line 853 below : ------------------------------------------------------- old_sock = rtp_session->sock_input; rtp_session->sock_input = new_sock; new_sock = NULL; if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER) || switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK)) { switch_socket_opt_set(rtp_session->sock_input, SWITCH_SO_NONBLOCK, TRUE); switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_NOBLOCK); } switch_socket_create_pollset(&rtp_session->read_pollfd, rtp_session->sock_input, SWITCH_POLLIN | SWITCH_POLLERR, rtp_session->pool); status = SWITCH_STATUS_SUCCESS; *err = "Success"; switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_IO); done: if (new_sock) { switch_socket_close(new_sock); } if (old_sock) { switch_socket_close(old_sock); } if (rtp_session->ready != 1) { WRITE_DEC(rtp_session); READ_DEC(rtp_session); } return status; ----------------------------------- it seems that old_sock is the actual sock that it needs to use since there is no auto changing to do ? So does it mean maybe the old_sock is yet existing because there is no NAT ? Thanks ----- Original Message ----- From: "Madovsky" <infos@madovsky.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Monday, March 22, 2010 2:31 AM Subject: Re: [Freeswitch-users] rtp timeout yet
I deactivated iptables, no router, no nat, only 2 public IP, 2 softphones connected on the same FS which is on a public IP to. default config, result -> rtp timeout crazy that's not working since I can make 2 softphones behind a firewall work....
ok that's enough for me tonight ! ;)
----- Original Message ----- From: "Madovsky" <infos@madovsky.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Monday, March 22, 2010 1:55 AM Subject: Re: [Freeswitch-users] rtp timeout yet
I replace public ip by these fake one...
----- Original Message ----- From: "Jason White" <jason@jasonjgw.net> To: <freeswitch-users@lists.freeswitch.org> Sent: Monday, March 22, 2010 1:49 AM Subject: Re: [Freeswitch-users] rtp timeout yet
Madovsky <infos@madovsky.org> wrote:
result with tcpdump -ennqti br0 -T rtp -vvv dst port 5060
10.10.10.132 = softphone1 registered from public IP
11.11.11.= softphone2 registered from public IP
10.10.10.138 = FS
Those are private addresses; there's a NAT device involved somewhere, and that's likely to be where your problems are.
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Madovsky <infos@madovsky.org> wrote:
to be sure I'm silly I downgraded from trunk 17001 and rtp timeout disappeared.
Try the latest trunk. If the problem still occurs, capture SIP traces and submit a bug report, making sure the configurations are identical.
thanks for your answer Jason. yes the problem is yet there. before to submit a bug I want to be sure it's a mistake I did. so when I set inbound-bypass-media to true, the problem disappear. it's for sure a media problem. A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp timeout A phone behind a firewall (tested on 6 PCs too) registers well and doesn't make rtp timeout. I stopped all IP rules like iptables etc... without success. I'd like to understand Thanks ----- Original Message ----- From: "Jason White" <jason@jasonjgw.net> To: <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, March 23, 2010 4:32 AM Subject: Re: [Freeswitch-users] rtp timeout yet
Madovsky <infos@madovsky.org> wrote:
to be sure I'm silly I downgraded from trunk 17001 and rtp timeout disappeared.
Try the latest trunk. If the problem still occurs, capture SIP traces and submit a bug report, making sure the configurations are identical.
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did you get a pcap trace of the file with tshark or tcpdump? also consider doing these cli command and reproducing console loglevel debug sofia profile <profname> siptrace on (substitute <profname> for your profile name run it on every profile) also look at sofia status profile <profname> and see what the output of that says. you could post it on pastebin if you need help understanding what it means On Tue, Mar 23, 2010 at 11:26 AM, Madovsky <infos@madovsky.org> wrote:
thanks for your answer Jason. yes the problem is yet there. before to submit a bug I want to be sure it's a mistake I did. so when I set inbound-bypass-media to true, the problem disappear. it's for sure a media problem. A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp timeout A phone behind a firewall (tested on 6 PCs too) registers well and doesn't make rtp timeout. I stopped all IP rules like iptables etc... without success. I'd like to understand
Thanks
----- Original Message ----- From: "Jason White" <jason@jasonjgw.net> To: <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, March 23, 2010 4:32 AM Subject: Re: [Freeswitch-users] rtp timeout yet
Madovsky <infos@madovsky.org> wrote:
to be sure I'm silly I downgraded from trunk 17001 and rtp timeout disappeared.
Try the latest trunk. If the problem still occurs, capture SIP traces and submit a bug report, making sure the configurations are identical.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I am having a similar problem with SIP UA based on PJSIP. My main issue relate with sometimes having one-way audio between two peers. Exactly the same peers sometimes works, sometime does not work. What i see is that when everything works are two statement in debug log: [INFO] switch_rtp.c:2005 Auto Changing port from ... [INFO] switch_rtp.c:2005 Auto Changing port from ... When things does not work (one-way audio) i just see 1 line: [INFO] switch_rtp.c:2005 Auto Changing port from ... For all calls i am using inbound-proxy-media=true . Is the same issue you see/encounter? Fabio On 23/03/10 17.26, Madovsky wrote:
thanks for your answer Jason. yes the problem is yet there. before to submit a bug I want to be sure it's a mistake I did. so when I set inbound-bypass-media to true, the problem disappear. it's for sure a media problem. A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp timeout A phone behind a firewall (tested on 6 PCs too) registers well and doesn't make rtp timeout. I stopped all IP rules like iptables etc... without success. I'd like to understand
Thanks
That message is a symptom of the UA lying in it's sdp. it says one ip in the sdp but means another. We start by sending all the media to the ip specified in the SDP because that is the only IP we know. Then there is a small window at the beginning of the call where we assume a packet sent to us where we specified in our SDP may be a packet from a mis-configured UA, so we will change our destination to that ip:port to defeat the bug in the client. in order to engage this fix the client has to send us several packets so we can prove it's correct to change our target towards them. On Tue, Mar 23, 2010 at 11:39 AM, Fabio Pietrosanti (naif) < lists@infosecurity.ch> wrote:
I am having a similar problem with SIP UA based on PJSIP.
My main issue relate with sometimes having one-way audio between two peers. Exactly the same peers sometimes works, sometime does not work.
What i see is that when everything works are two statement in debug log: [INFO] switch_rtp.c:2005 Auto Changing port from ... [INFO] switch_rtp.c:2005 Auto Changing port from ...
When things does not work (one-way audio) i just see 1 line: [INFO] switch_rtp.c:2005 Auto Changing port from ...
For all calls i am using inbound-proxy-media=true .
Is the same issue you see/encounter?
Fabio
On 23/03/10 17.26, Madovsky wrote:
thanks for your answer Jason. yes the problem is yet there. before to submit a bug I want to be sure it's a mistake I did. so when I set inbound-bypass-media to true, the problem disappear. it's for sure a media problem. A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp timeout A phone behind a firewall (tested on 6 PCs too) registers well and doesn't make rtp timeout. I stopped all IP rules like iptables etc... without success. I'd like to understand
Thanks
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hi, now i start to better understand. I am not using STUN or ICE and in my case clients are in 2 situations: a) The client has a private ip address between a NAT Port address transalation b) The client has a public ip address but that address is firewalled (3G operator that provide a public IP address but does not let incoming connection to go trough). In that scenario i got one-way audio with only one "Auto changing port from ..." messages from FS debug log. Is there a way to force the FS not to look at the IP inside the SDP header provisioned by the SIP UA, considering the IP address always "buggy", so always not providing the right IP address configuration ? Fabio On 23/03/10 17.51, Anthony Minessale wrote:
That message is a symptom of the UA lying in it's sdp. it says one ip in the sdp but means another.
We start by sending all the media to the ip specified in the SDP because that is the only IP we know. Then there is a small window at the beginning of the call where we assume a packet sent to us where we specified in our SDP may be a packet from a mis-configured UA, so we will change our destination to that ip:port to defeat the bug in the client.
in order to engage this fix the client has to send us several packets so we can prove it's correct to change our target towards them.
On Tue, Mar 23, 2010 at 11:39 AM, Fabio Pietrosanti (naif) <lists@infosecurity.ch <mailto:lists@infosecurity.ch>> wrote:
I am having a similar problem with SIP UA based on PJSIP.
My main issue relate with sometimes having one-way audio between two peers. Exactly the same peers sometimes works, sometime does not work.
What i see is that when everything works are two statement in debug log: [INFO] switch_rtp.c:2005 Auto Changing port from ... [INFO] switch_rtp.c:2005 Auto Changing port from ...
When things does not work (one-way audio) i just see 1 line: [INFO] switch_rtp.c:2005 Auto Changing port from ...
For all calls i am using inbound-proxy-media=true .
Is the same issue you see/encounter?
Fabio
On 23/03/10 17.26, Madovsky wrote: > thanks for your answer Jason. > yes the problem is yet there. before > to submit a bug I want to be sure it's a mistake I did. > so when I set inbound-bypass-media to true, the problem disappear. > it's for sure a media problem. > A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp > timeout > A phone behind a firewall (tested on 6 PCs too) registers well and doesn't > make rtp timeout. > I stopped all IP rules like iptables etc... without success. > I'd like to understand > > Thanks >
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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No we don't have such a feature because the media and signaling are separated by design of SIP. It would be a horrible assumption to always think the media was coming from the same IP as the SIP messages. And even then you still need the port # On Tue, Mar 23, 2010 at 1:00 PM, Fabio Pietrosanti (naif) < lists@infosecurity.ch> wrote:
Hi,
now i start to better understand.
I am not using STUN or ICE and in my case clients are in 2 situations: a) The client has a private ip address between a NAT Port address transalation b) The client has a public ip address but that address is firewalled (3G operator that provide a public IP address but does not let incoming connection to go trough).
In that scenario i got one-way audio with only one "Auto changing port from ..." messages from FS debug log.
Is there a way to force the FS not to look at the IP inside the SDP header provisioned by the SIP UA, considering the IP address always "buggy", so always not providing the right IP address configuration ?
Fabio
On 23/03/10 17.51, Anthony Minessale wrote:
That message is a symptom of the UA lying in it's sdp. it says one ip in the sdp but means another.
We start by sending all the media to the ip specified in the SDP because that is the only IP we know. Then there is a small window at the beginning of the call where we assume a packet sent to us where we specified in our SDP may be a packet from a mis-configured UA, so we will change our destination to that ip:port to defeat the bug in the client.
in order to engage this fix the client has to send us several packets so we can prove it's correct to change our target towards them.
On Tue, Mar 23, 2010 at 11:39 AM, Fabio Pietrosanti (naif) < lists@infosecurity.ch> wrote:
I am having a similar problem with SIP UA based on PJSIP.
My main issue relate with sometimes having one-way audio between two peers. Exactly the same peers sometimes works, sometime does not work.
What i see is that when everything works are two statement in debug log: [INFO] switch_rtp.c:2005 Auto Changing port from ... [INFO] switch_rtp.c:2005 Auto Changing port from ...
When things does not work (one-way audio) i just see 1 line: [INFO] switch_rtp.c:2005 Auto Changing port from ...
For all calls i am using inbound-proxy-media=true .
Is the same issue you see/encounter?
Fabio
On 23/03/10 17.26, Madovsky wrote:
thanks for your answer Jason. yes the problem is yet there. before to submit a bug I want to be sure it's a mistake I did. so when I set inbound-bypass-media to true, the problem disappear. it's for sure a media problem. A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp timeout A phone behind a firewall (tested on 6 PCs too) registers well and doesn't make rtp timeout. I stopped all IP rules like iptables etc... without success. I'd like to understand
Thanks
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
So which is the right way to "stimulate" FS to detect that condition quickly? To send out RTP packets as early as possible from the SIP UA? What i am still missing is if FS will always rewrite the SDP header to act as a media relay between the caller and the called or not, so forcing the clients to only send packets to FS. Fabio On 23/03/10 19.09, Anthony Minessale wrote:
No we don't have such a feature because the media and signaling are separated by design of SIP. It would be a horrible assumption to always think the media was coming from the same IP as the SIP messages. And even then you still need the port #
yes you have to start sending the rtp to FS when you are given the SDP FS can only detect the error and correct it if it gets several RTP packets. the other option is to use stun on the client so it can put the right ip in the SDP to begin with. On Tue, Mar 23, 2010 at 2:10 PM, Fabio Pietrosanti (naif) < lists@infosecurity.ch> wrote:
So which is the right way to "stimulate" FS to detect that condition quickly?
To send out RTP packets as early as possible from the SIP UA?
What i am still missing is if FS will always rewrite the SDP header to act as a media relay between the caller and the called or not, so forcing the clients to only send packets to FS.
Fabio
On 23/03/10 19.09, Anthony Minessale wrote:
No we don't have such a feature because the media and signaling are separated by design of SIP. It would be a horrible assumption to always think the media was coming from the same IP as the SIP messages. And even then you still need the port #
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
no I don't use proxy ----- Original Message ----- From: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch> To: <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, March 23, 2010 12:39 PM Subject: Re: [Freeswitch-users] rtp timeout yet
I am having a similar problem with SIP UA based on PJSIP.
My main issue relate with sometimes having one-way audio between two peers. Exactly the same peers sometimes works, sometime does not work.
What i see is that when everything works are two statement in debug log: [INFO] switch_rtp.c:2005 Auto Changing port from ... [INFO] switch_rtp.c:2005 Auto Changing port from ...
When things does not work (one-way audio) i just see 1 line: [INFO] switch_rtp.c:2005 Auto Changing port from ...
For all calls i am using inbound-proxy-media=true .
Is the same issue you see/encounter?
Fabio
On 23/03/10 17.26, Madovsky wrote:
thanks for your answer Jason. yes the problem is yet there. before to submit a bug I want to be sure it's a mistake I did. so when I set inbound-bypass-media to true, the problem disappear. it's for sure a media problem. A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp timeout A phone behind a firewall (tested on 6 PCs too) registers well and doesn't make rtp timeout. I stopped all IP rules like iptables etc... without success. I'd like to understand
Thanks
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participants (4)
-
Anthony Minessale
-
Fabio Pietrosanti (naif)
-
Jason White
-
Madovsky