[Freeswitch-dev] FS send re-invite about every minutes, is there a way to disable it?
Hi, The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description". I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it? btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.
Seven, This is called session timers, its sent out to make sure the call is still active. Since SIP is UDP-based, if the UA is unplugged, the call would just stay active. You can look in your UA's settings to see if you can make that longer or disable it. If you do a SIP trace, look for the "Session-Expires" header. (Should be like: Session-Expires: <nbr of seconds>). When FS performs an outbound call, this is controlled by the session- timeout parameter in the sip profile. Eg: <param name="session-timeout" value="120"/> ~Math On 26-Feb-09, at 10:12 PM, seven wrote:
Hi,
The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description".
I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it?
btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Thanks, but doesn't OPTION msg can make sure KEEP-ALIVE? I have no way to edit the bad UA settings for Session-Expires. But I traced SIP and there is no Session-Expires in the UA's INVITE msg, I guess FS default that to 1 minute so it send out the re-INVITE. And INVITE sent by FS do has a Session-Expires:120 header . Seven- On Feb 27, 2009, at 11:23 AM, Mathieu Rene wrote:
Seven,
This is called session timers, its sent out to make sure the call is still active. Since SIP is UDP-based, if the UA is unplugged, the call would just stay active.
You can look in your UA's settings to see if you can make that longer or disable it.
If you do a SIP trace, look for the "Session-Expires" header. (Should be like: Session-Expires: <nbr of seconds>).
When FS performs an outbound call, this is controlled by the session- timeout parameter in the sip profile. Eg: <param name="session-timeout" value="120"/>
~Math
On 26-Feb-09, at 10:12 PM, seven wrote:
Hi,
The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description".
I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it?
btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
OPTIONS messages arent call-related, they can be used to check connectivity but is not relevant in the event of a call being proxied. Could you attach a sip trace? and which UA are we talking about? ~Math On 26-Feb-09, at 11:05 PM, seven wrote:
Thanks, but doesn't OPTION msg can make sure KEEP-ALIVE?
I have no way to edit the bad UA settings for Session-Expires. But I traced SIP and there is no Session-Expires in the UA's INVITE msg, I guess FS default that to 1 minute so it send out the re-INVITE. And INVITE sent by FS do has a Session-Expires:120 header .
Seven-
On Feb 27, 2009, at 11:23 AM, Mathieu Rene wrote:
Seven,
This is called session timers, its sent out to make sure the call is still active. Since SIP is UDP-based, if the UA is unplugged, the call would just stay active.
You can look in your UA's settings to see if you can make that longer or disable it.
If you do a SIP trace, look for the "Session-Expires" header. (Should be like: Session-Expires: <nbr of seconds>).
When FS performs an outbound call, this is controlled by the session- timeout parameter in the sip profile. Eg: <param name="session-timeout" value="120"/>
~Math
On 26-Feb-09, at 10:12 PM, seven wrote:
Hi,
The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description".
I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it?
btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Thank you. I first noticed this and posted on JIRA: http://jira.freeswitch.org/browse/SFSIP-110 , however it's another problem(network problem I think). The easiest way to make a sip trace is from our production system and I it's not good to post in public, hope email you off list is OK. On Feb 27, 2009, at 12:09 PM, Mathieu Rene wrote:
OPTIONS messages arent call-related, they can be used to check connectivity but is not relevant in the event of a call being proxied.
Could you attach a sip trace? and which UA are we talking about?
~Math
On 26-Feb-09, at 11:05 PM, seven wrote:
Thanks, but doesn't OPTION msg can make sure KEEP-ALIVE?
I have no way to edit the bad UA settings for Session-Expires. But I traced SIP and there is no Session-Expires in the UA's INVITE msg, I guess FS default that to 1 minute so it send out the re-INVITE. And INVITE sent by FS do has a Session-Expires:120 header .
Seven-
On Feb 27, 2009, at 11:23 AM, Mathieu Rene wrote:
Seven,
This is called session timers, its sent out to make sure the call is still active. Since SIP is UDP-based, if the UA is unplugged, the call would just stay active.
You can look in your UA's settings to see if you can make that longer or disable it.
If you do a SIP trace, look for the "Session-Expires" header. (Should be like: Session-Expires: <nbr of seconds>).
When FS performs an outbound call, this is controlled by the session- timeout parameter in the sip profile. Eg: <param name="session-timeout" value="120"/>
~Math
On 26-Feb-09, at 10:12 PM, seven wrote:
Hi,
The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description".
I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it?
btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
FS 200 OKs with a session-timeout header, add that to your sip profile <param name="session-timeout" value="0"/> It will disable them. ~Math On 27-Feb-09, at 1:25 AM, seven wrote:
Thank you.
I first noticed this and posted on JIRA: http://jira.freeswitch.org/browse/SFSIP-110 , however it's another problem(network problem I think). The easiest way to make a sip trace is from our production system and I it's not good to post in public, hope email you off list is OK.
On Feb 27, 2009, at 12:09 PM, Mathieu Rene wrote:
OPTIONS messages arent call-related, they can be used to check connectivity but is not relevant in the event of a call being proxied.
Could you attach a sip trace? and which UA are we talking about?
~Math
On 26-Feb-09, at 11:05 PM, seven wrote:
Thanks, but doesn't OPTION msg can make sure KEEP-ALIVE?
I have no way to edit the bad UA settings for Session-Expires. But I traced SIP and there is no Session-Expires in the UA's INVITE msg, I guess FS default that to 1 minute so it send out the re-INVITE. And INVITE sent by FS do has a Session-Expires:120 header .
Seven-
On Feb 27, 2009, at 11:23 AM, Mathieu Rene wrote:
Seven,
This is called session timers, its sent out to make sure the call is still active. Since SIP is UDP-based, if the UA is unplugged, the call would just stay active.
You can look in your UA's settings to see if you can make that longer or disable it.
If you do a SIP trace, look for the "Session-Expires" header. (Should be like: Session-Expires: <nbr of seconds>).
When FS performs an outbound call, this is controlled by the session- timeout parameter in the sip profile. Eg: <param name="session-timeout" value="120"/>
~Math
On 26-Feb-09, at 10:12 PM, seven wrote:
Hi,
The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description".
I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it?
btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Hi Math, in 200 OK session-timeout, do you mean this? Session-Expires: 120;refresher=uas I added the session-timeout to my sip profile and I can see SESSION_TO 0 in console when I typed in " sofia status profile default " , however the Session-Expires will be always 120 in the 200OK msg. Oh, I'm behind a NAT, so I think the timeout always equals 20 (mod_sofia.c around line 519: session_timeout = SOFIA_NAT_SESSION_TIMEOUT; ) I tried change <param name="minimum-session-expires" value="0"/> and I can get Session-Expires: 90;refresher=uas Min-SE: 90 However, I still get the re-INVITE message after 60 seconds. On Feb 27, 2009, at 10:31 PM, Mathieu Rene wrote:
FS 200 OKs with a session-timeout header, add that to your sip profile
<param name="session-timeout" value="0"/>
It will disable them.
~Math
On 27-Feb-09, at 1:25 AM, seven wrote:
Thank you.
I first noticed this and posted on JIRA: http://jira.freeswitch.org/browse/SFSIP-110 , however it's another problem(network problem I think). The easiest way to make a sip trace is from our production system and I it's not good to post in public, hope email you off list is OK.
On Feb 27, 2009, at 12:09 PM, Mathieu Rene wrote:
OPTIONS messages arent call-related, they can be used to check connectivity but is not relevant in the event of a call being proxied.
Could you attach a sip trace? and which UA are we talking about?
~Math
On 26-Feb-09, at 11:05 PM, seven wrote:
Thanks, but doesn't OPTION msg can make sure KEEP-ALIVE?
I have no way to edit the bad UA settings for Session-Expires. But I traced SIP and there is no Session-Expires in the UA's INVITE msg, I guess FS default that to 1 minute so it send out the re-INVITE. And INVITE sent by FS do has a Session-Expires:120 header .
Seven-
On Feb 27, 2009, at 11:23 AM, Mathieu Rene wrote:
Seven,
This is called session timers, its sent out to make sure the call is still active. Since SIP is UDP-based, if the UA is unplugged, the call would just stay active.
You can look in your UA's settings to see if you can make that longer or disable it.
If you do a SIP trace, look for the "Session-Expires" header. (Should be like: Session-Expires: <nbr of seconds>).
When FS performs an outbound call, this is controlled by the session- timeout parameter in the sip profile. Eg: <param name="session-timeout" value="120"/>
~Math
On 26-Feb-09, at 10:12 PM, seven wrote:
Hi,
The recent version of FS( I think I had notice that also on 1.0.1 stable version) seems sends out an INVITE msg every minute when I originate a call from a SIPUA(say X-Lite), then the SIPUA will respond 200 OK with SDP msg. There's no problem with X-Lite but some bad SIPUA can send bad responding msg and will cause FS hangup the call said "Malformed Session Description".
I know it's not FS's problem, but I don't understand why it send the INVITE(a re-INVITE?) every minute? is there a way to disable it?
btw, it only happens when you originate a call from a SIPUA, I originate from FS(originate sofia/defaut/1000 &echo) doesn't get re- INVITE msg.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- dev http://www.freeswitch.org
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Seven, Do you know how Session timers work? If for example a session timer is setup at 120 seconds you'll get an invite about every 60 seconds. /b On Feb 28, 2009, at 1:24 AM, seven du wrote:
Hi Math,
in 200 OK session-timeout, do you mean this?
Session-Expires: 120;refresher=uas
I added the session-timeout to my sip profile and I can see
SESSION_TO 0
in console when I typed in " sofia status profile default " , however the Session-Expires will be always 120 in the 200OK msg.
Oh, I'm behind a NAT, so I think the timeout always equals 20 (mod_sofia.c around line 519: session_timeout = SOFIA_NAT_SESSION_TIMEOUT; )
I tried change
<param name="minimum-session-expires" value="0"/>
and I can get
Session-Expires: 90;refresher=uas Min-SE: 90
However, I still get the re-INVITE message after 60 seconds.
Thanks Brian. I didn't know that before. But doesn't set the timer to 0 disable it? Or can I set to a very large value say 9999999? Anyway, if I change session-timeout, should I see the difference at the sip response header "Session-Expires: 120;refresher=uas" right? Do you think it related to this? http://www.nabble.com/change-session-timer-value-td22239490.html On Mar 1, 2009, at 12:40 AM, Brian West wrote:
Seven, Do you know how Session timers work? If for example a session timer is setup at 120 seconds you'll get an invite about every 60 seconds.
/b
On Feb 28, 2009, at 1:24 AM, seven du wrote:
Hi Math,
in 200 OK session-timeout, do you mean this?
Session-Expires: 120;refresher=uas
I added the session-timeout to my sip profile and I can see
SESSION_TO 0
in console when I typed in " sofia status profile default " , however the Session-Expires will be always 120 in the 200OK msg.
Oh, I'm behind a NAT, so I think the timeout always equals 20 (mod_sofia.c around line 519: session_timeout = SOFIA_NAT_SESSION_TIMEOUT; )
I tried change
<param name="minimum-session-expires" value="0"/>
and I can get
Session-Expires: 90;refresher=uas Min-SE: 90
However, I still get the re-INVITE message after 60 seconds.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
That depends on if its inbound or outbound.. you have to pay close attention to the direction and uas vs uac. /b On Feb 28, 2009, at 12:32 PM, seven du wrote:
Thanks Brian. I didn't know that before. But doesn't set the timer to 0 disable it? Or can I set to a very large value say 9999999?
Anyway, if I change session-timeout, should I see the difference at the sip response header "Session-Expires: 120;refresher=uas" right?
Do you think it related to this? http://www.nabble.com/change-session-timer-value-td22239490.html
If you go look in sofia's code you see something like this: /* refresh 30..60 seconds before deadline */ /* By default, refresh around half time before deadline */ /b On Feb 28, 2009, at 12:32 PM, seven du wrote:
Thanks Brian. I didn't know that before. But doesn't set the timer to 0 disable it? Or can I set to a very large value say 9999999?
Anyway, if I change session-timeout, should I see the difference at the sip response header "Session-Expires: 120;refresher=uas" right?
Do you think it related to this?
Got it. because I'm behind an NAT and #define SOFIA_NAT_SESSION_TIMEOUT 20 515 if (switch_test_flag(tech_pvt, TFLAG_NAT) || 516 (val = switch_channel_get_variable(channel, "sip-force- contact")) || 517 ((val = switch_channel_get_variable(channel, "sip_sticky_contact")) && switch_true(val))) { 518 sticky = tech_pvt->record_route; 519 session_timeout = SOFIA_NAT_SESSION_TIMEOUT; so, session_timeout would always be 20 and I always get a Session- Expires header with value 120(the default value maybe). Based on http://tools.ietf.org/html/draft-ietf-sip-session-timer-15, I think #define SOFIA_NAT_SESSION_TIMEOUT 20 maybe not followed the standard. Anyway, OPTION packets works well for NAT KEEP-ALIVE right? Do we really need line 519(as the above code)? or should we make another setting parameter say "nat_session_timeout" in sofia profile? Do I need to post this on jira? On Mar 1, 2009, at 4:19 AM, Brian West wrote:
If you go look in sofia's code you see something like this:
/* refresh 30..60 seconds before deadline */
/* By default, refresh around half time before deadline */
/b
On Feb 28, 2009, at 12:32 PM, seven du wrote:
Thanks Brian. I didn't know that before. But doesn't set the timer to 0 disable it? Or can I set to a very large value say 9999999?
Anyway, if I change session-timeout, should I see the difference at the sip response header "Session-Expires: 120;refresher=uas" right?
Do you think it related to this?
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
participants (4)
-
Brian West
-
Mathieu Rene
-
seven
-
seven du