[Freeswitch-users] Require Timer on re-invite from FS to A-LEG
Anthony Minessale
anthony.minessale at gmail.com
Mon Sep 17 19:11:25 MSD 2012
Just because it says its not recommended does not mean it says you should
intentionally drop a call because of it.
If you can make it work by turning off in the var then that is your
solution. As I have already said, there are several real-world situations
where it breaks things going either way hence our solution was to make it
work both ways.
On Mon, Sep 17, 2012 at 10:00 AM, Mike Burlingame <mike.burlingame at me.com>wrote:
> PaeTec and Global Crossing both feel that we should not be including the
> require based on the RFC posted below - I have read over the RFC and tend
> to agree based on my limited knowledge of the timer RFC
>
> Based on what they are saying because the invite from them state that
> timers are supported that we should remember this and not be putting in a
> require
>
> ------------------
> From carriers
> -------------------
>
> Please see below…section 8.1 of RFC 4028****
>
>
>
> From RFC4028:****
>
>
>
> The proxy MUST remember, for the duration of the transaction, whether**
> **
>
> the request contained the Supported header field with the value****
>
> 'timer'. If the request did not contain a Supported header field****
>
> with the value 'timer', the proxy *MAY* insert a Require header field**
> **
>
> with the value 'timer' into the request. However, this is*NOT****
>
> RECOMMENDED*. This allows the proxy to insist on a session timer for**
> **
>
> the session. This header field is not needed if a Supported header****
>
> field was in the request; in this case, the proxy would already be****
>
> sure the session timer can be used for the session.****
>
>
>
>
>
> Section 7.1 of RFC 4028****
>
>
>
> A UAC that supports the session timer extension defined here MUST****
>
> include a Supported header field in each request (except ACK),****
>
> listing the option tag 'timer' [2<http://tools.ietf.org/html/rfc4028#ref-2>].
> It MUST do so even if the UAC is****
>
> not requesting usage of the session timer for this session.****
>
>
>
> The UAC MAY include a Require header field in the request with the****
>
> value 'timer' to indicate that the UAS must support the session timer**
> **
>
> to participate in the session. This does not mean that the UAC is****
>
> requiring the UAS to perform the refreshes, only that it is requiring**
> **
>
> the UAS to support the extension. In addition, the UAC MAY include a**
> **
>
> Proxy-Require header field in the request with the value 'timer' to****
>
> indicate that proxies must support the session timer in order to****
>
> correctly process the request. However, usage of either Require or****
>
> Proxy-Require by the UAC is NOT RECOMMENDED. They are not needed,****
>
> since the extension works even when only the UAC supports the****
>
> extension. The Supported header field containing 'timer' MUST still***
> *
>
> be included, even if the Require or Proxy-Require header fields are****
>
> present containing 'timer'****
>
>
>
> Sent from my iPhone 4S
>
> On Sep 14, 2012, at 5:29 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> It's Wong to not accept it. IIRC...there is a better case for why it
> should be there than for it not being there. It's a rather dumb thing for
> some of these commercial switches to even care about....I don't want to set
> a precedence here that I will perpetually change our sip stack and chase
> after every interop case... Make sure the param is set, I believe it
> disables it completely going against the rfc but I can't check till
> Monday. This is part of Sofia not FS itself.
> On Sep 14, 2012 7:19 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
>
>> Thanks that seem to help the high session timer value however the
>> require: timer is still present
>>
>> Invite from B-Leg --> FreeSwitch
>> recv 913 bytes from udp/[Outbound_Carrier_Proxy]:5060 at 23:50:28.717807:
>>
>> ------------------------------------------------------------------------
>> INVITE sip:18665551212 at FreeSwitch:5060 SIP/2.0
>> Via: SIP/2.0/UDP Outbound_Carrier_Proxy;branch=z9hG4bKcb3e.af955752.0
>> Via: SIP/2.0/UDP
>> Carrier_IP:5060;branch=z9hG4bK80qcmt0080p0ntsfn2j0sbd83gel1.1
>> Allow-Events: message-summary, refer, dialog, line-seize, presence,
>> call-info, as-feature-event
>> Max-Forwards: 68
>> Call-ID: 28526ec2-4570-428b-960e-a065b7fcdfac
>> From: <sip:18665551212 at Carrier_IP>;tag=10.152.0.77+1+63320+3c304bf3
>> To: "BURLINGAME MIKE" <sip:13605551212 at X.X.X.X>;tag=B688eNFc2yv3e
>> CSeq: 219015274 INVITE
>> Expires: 180
>> Contact: <sip:19999218665551212 at Carrier_IP:5060;transport=udp>
>> Organization: MetaSwitch
>> Supported: 100rel, resource-priority
>> Content-Length: 193
>> Content-Type: application/sdp
>>
>> v=0
>> o=- 3341834898 3341834898 IN IP4 Carrier_IP
>> s=-
>> c=IN IP4 Carrier_IP
>> t=0 0
>> m=audio 29864 RTP/AVP 18 0 101
>> a=rtpmap:101 telephone-event/8000
>> a=ptime:20
>> a=silenceSupp:off - - - -
>>
>>
>>
>> Invite from FreeSwitch --> A-Leg
>> send 1000 bytes to udp/[ULC]:5060 at 23:50:28.718781:
>>
>> ------------------------------------------------------------------------
>> INVITE sip:mod_sofia at A-LEG_IP:5070 SIP/2.0
>> Via: SIP/2.0/UDP FreeSwitch;rport;branch=z9hG4bKjFHtgQ8r4p0Sg
>> Route: <sip:ULC;lr=on>
>> Max-Forwards: 47
>> From: <sip:18665551212 at ULC>;tag=aXFgDty84N6gK
>> To: "BURLINGAME MIKE" <sip:13605551212 at A-LEG_IP>;tag=XeZZ4FrX4gXeg
>> Call-ID: 71f58b05-5d2c-4f09-a390-66bf9698eff5
>> CSeq: 33497250 INVITE
>> Contact: <sip:18665551212 at FreeSwitch:5060;transport=udp>
>> User-Agent: FreeSwitch
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
>> Require: timer
>> Supported: timer, precondition, path, replaces
>> Session-Expires: 3600;refresher=uas
>> Min-SE: 120
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 193
>> P-hint: outbound
>> X-FS-Support: update_display,send_info
>>
>> v=0
>> o=- 3341834898 3341834898 IN IP4 Carrier_IP
>> s=-
>> c=IN IP4 Carrier_IP
>> t=0 0
>> m=audio 29864 RTP/AVP 18 0 101
>> a=rtpmap:101 telephone-event/8000
>> a=ptime:20
>> a=silenceSupp:off - - - -
>>
>>
>> On Sep 14, 2012, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> Mike, try comma separating the values instead of using two sets of {}:
>>
>> add_action("bridge","{sip_wait_for_aleg_ack=true,sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>
>> -MC
>>
>> On Fri, Sep 14, 2012 at 3:38 PM, Mike Burlingame <mike.burlingame at me.com>wrote:
>>
>>> I have that set in the bridge command however does not seem to be
>>> working as expected.
>>>
>>>
>>> add_action("bridge","{sip_wait_for_aleg_ack=true}{sip_require_timer=false}[sip_contact_user=$ext,sip_auth_username=".$tp_username.",sip_auth_password=".$tp_password.$x_lrn."]sofia/fs/".$tp_trunkprefix.$ext."@".$tp_providerip.";fs_path=sip:OUT_PROXY");
>>>
>>>
>>>
>>> On Sep 14, 2012, at 3:16 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>> set {sip_require_timer=false} in your outbound calls or globally
>>>
>>>
>>> On Fri, Sep 14, 2012 at 5:10 PM, Mike Burlingame <mike.burlingame at me.com
>>> > wrote:
>>>
>>>> it seems if I get an re-invite from the B-Leg FS add's requires timer
>>>> and changes the session timer to a high value to the re-invite going to the
>>>> A-Leg come to find out Acme Packets at our ULC's do not like this and send
>>>> us back a 420 Bad Extension and the call is disconnected with Reason:
>>>> Q.850;cause=127;text="INTERWORKING"
>>>>
>>>> The ULC are stating we need to only have it in our supported and do not
>>>> pass them a require.
>>>>
>>>> Re-Invite from B-Leg to FS
>>>>
>>>> ------------------------------------------------------------------------
>>>> INVITE sip:16025551212;phone-context=+1 at FS_SERVER:5070 SIP/2.0
>>>> Via: SIP/2.0/UDP OpenSIPs_DID_Proxy;branch=z9hG4bK8288.10e55fe5.0
>>>> Via: SIP/2.0/UDP
>>>> B-LEG_IP:5060;branch=z9hG4bKve7vp6002gr1gfsh72k0sb1gv1ek1.1
>>>> Call-Id: f8330599-d298-4233-92bb-5d622e85aa6d
>>>> Contact: <sip:18475551212 at B-LEG_IP:5060;transport=udp>
>>>> Content-Length: 217
>>>> Content-Type: application/sdp
>>>> CSeq: 33480808 INVITE
>>>> From: <sip:18475551212 at B-LEG_IP>;tag=100052073
>>>> Max-Forwards: 92
>>>> Session-Expires: 3600;refresher=uas
>>>> Supported: timer
>>>> To: <sip:116025551212 at OpenSIPs_DID_Proxy>;tag=Dj92X5t8065FQ
>>>> User-Agent: FreeSwitch
>>>>
>>>> v=0
>>>> o=- 3308986892 0 IN IP4 B-LEG_IP
>>>> s=Media Server
>>>> c=IN IP4 B-LEG_IP
>>>> t=0 0
>>>> m=audio 51246 RTP/AVP 0 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>> Re-Invite from FS to A-Leg
>>>>
>>>> ------------------------------------------------------------------------
>>>> INVITE sip:16025551212 at DID_CARRIER:5060;transport=udp SIP/2.0
>>>> Via: SIP/2.0/UDP FS_SERVER:5070;rport;branch=z9hG4bKj8yevKKS0X0mK
>>>> Route: <sip:OpenSIPS_DID_CARRIER_Proxy;lr=on>
>>>> Max-Forwards: 97
>>>> From: <sip:18475551212;phone-context=+1 at OpenSIPS_DID_CARRIER_Proxy
>>>> :5060;user=phone>;tag=c9Favaa53XFXB
>>>> To: <sip:16025551212;phone-context=+1 at DID_CARRIER
>>>> :5060;user=phone>;tag=SDd626401-gK095bbb72
>>>> Call-ID: SDd626401-b06f59c4aa359462042e25890d7b5bfd-v3000i1
>>>> CSeq: 33480811 INVITE
>>>> Contact: <
>>>> sip:18475551212;phone-context=+1 at FS_SERVER:5070;transport=udp>
>>>> User-Agent: FreeSwitch
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> NOTIFY
>>>> Require: timer
>>>> Supported: timer, precondition, path, replaces
>>>> Session-Expires: 64800;refresher=uas
>>>> Min-SE: 64800
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 235
>>>> X-FS-Support: update_display,send_info
>>>>
>>>> v=0
>>>> o=- 3308979701 3213293310682935904 IN IP4 B-LEG_IP
>>>> s=Media Server
>>>> c=IN IP4 B-LEG_IP
>>>> t=0 0
>>>> m=audio 51246 RTP/AVP 0 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
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>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
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>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
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>>>
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>>>
>>>
>>>
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>>>
>>> _________________________________________________________________________
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>>>
>>>
>>>
>>>
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>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
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>> http://www.OSTAG.org <http://www.ostag.org/>
>>
>>
>> _________________________________________________________________________
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>>
>>
>>
>>
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>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
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>>
>>
>>
>>
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>> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
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>
> _________________________________________________________________________
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>
>
>
>
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com
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