[Freeswitch-users] Require Timer on re-invite from FS to A-LEG

Mike Burlingame mike.burlingame at me.com
Mon Sep 17 19:17:39 MSD 2012 

I agree they should not be dropping the call - was just passing on the info to complete out the thread as well as show it seems to be losing battle with the carriers to try to get them to change

Sent from my iPhone 4S

On Sep 17, 2012, at 8:11 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> 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].  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
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
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>>>>>>>> 
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>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> -- 
>>>>>>> Anthony Minessale II
>>>>>>> 
>>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>>> ClueCon http://www.cluecon.com/
>>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>> 
>>>>>>> AIM: anthm
>>>>>>> MSN:anthony_minessale at hotmail.com
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>> 
>>>>>>> FreeSWITCH Developer Conference
>>>>>>> sip:888 at conference.freeswitch.org
>>>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>>>> pstn:+19193869900
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
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>>>>>>> 
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>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
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>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> 
>>>>> -- 
>>>>> Michael S Collins
>>>>> Twitter: @mercutioviz
>>>>> http://www.FreeSWITCH.org
>>>>> http://www.ClueCon.com
>>>>> http://www.OSTAG.org
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
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>>>>> 
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>>>> 
>>>> 
>>>> _________________________________________________________________________
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>>>> 
>>>> 
>>>> 
>>>> 
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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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> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
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> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> 
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