Hi andres

these are the invites traces

Incoming INVITE from DID provider

Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:521234567890@kamailioServer:8080;user=phone SIP/2.0
    Message Header
        Via: SIP/2.0/UDP providerServer:5060;branch=z9hG4bK+5bef3865b99c5762a2c30f01ddd636b81+sip+1+a741b93a
        From: <sip:521584126220038@providerServer:5060>;tag=providerServer+1+371535df+e2ef11e9
        To: <sip:521234567890@kamailioServer:8080;user=phone>
        CSeq: 1 INVITE
        Expires: 180
        Content-Length: 341
        Call-Info: <sip:providerServer:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
        Supported: replaces,unknown, 100rel
        Contact: <sip:521584126220038@providerServer:5060;transport=udp>
        Content-Type: application/sdp
        Call-ID: 0gQAAC8WAAACBAAALxYAABgr6m9ER9mS85Wc6XubKwYpvcZXSrL4Nof25OH5ft1No6QQRSEQSjd66WwF/AS2zw--@providerServer
        Max-Forwards: 69
        Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,REFER,UPDATE
        User-Agent: Softphone
        Accept: application/sdp, application/dtmf-relay
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 1245688524974 1245688524974 IN IP4 providerServer
            Session Name (s): -
            Connection Information (c): IN IP4 providerServer
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 22602 RTP/AVP 8 0 96 18 101
            Media Attribute (a): rtpmap:96 G.729b/8000/1
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15
            Media Attribute (a): sqn:0
            Media Attribute (a): cdsc:1 audio RTP/AVP 100
            Media Attribute (a): cpar:a=rtpmap:100 X-NSE/8000
            Media Attribute (a): cpar:a=fmtp:100 192-194,200-202
            Media Attribute (a): cdsc:2 image udptl t38

INVITE redirect by kamailio

Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:U1234567890@192.168.0.101:7993 SIP/2.0

    Message Header
        Record-Route: <sip:kamailioServer:8080;lr=on;ftag=providerServer+1+371535df+e2ef11e9;nat=yes>
        User-Agent: softphone
        Supported: replaces
        Via: SIP/2.0/UDP kamailioServer:8080;branch=z9hG4bK2ec6.a16e6fb8fc4e536d16bcb018eeb36b28.1
        Via: SIP/2.0/UDP 10.20.82.230;branch=z9hG4bKsr-gJeXVzfKDCu6W9BODQBUVQNJpv11RC18Rz7MDjTapwqaIwqJbmgaRQBt6NOuhHeipiq8TksQljM4pkSxRHqV0z6dhkTzpjTMTQGsTzgwRQqSDoDzDHTaDkWGhjTzRof1Dv.zlPBrDv.SRzXUTQGzTX**
        From: <sip:521584126220038@providerServer:5060>;tag=providerServer+1+371535df+e2ef11e9
        To: <sip:521234567890@kamailioServer:8080;user=phone>
        CSeq: 1 INVITE
        Expires: 180
        Content-Length: 361
        Contact: <sip:10.20.82.230;line=sr-ZieapQg8Dmg1RjN8RQf8DjBzpNB8DjNKDmXsVQOJVQfwpQgaRQBtAxqSIFRaIwqJbP6GZj.SIHeSZzJ8DjNKDmXsVQOJVQfwLQgaRQYyDX**>
        Call-ID: 0gQAAC8WAAACBAAALxYAABgr6m9ER9mS85Wc6XubKwYpvcZXSrL4Nof25OH5ft1No6QQRSEQSjd66WwF/AS2zw--@providerServer
        Max-Forwards: 68
        Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,REFER,UPDATE
        Accept: application/sdp, application/dtmf-relay
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 1245688524974 1245688524974 IN IP4 208.101.36.104
            Session Name (s): -
            Connection Information (c): IN IP4 208.101.36.104
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 10958 RTP/AVP 8 0 96 18 101
            Media Attribute (a): rtpmap:96 G.729b/8000/1
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15
            Media Attribute (a): sqn:0
            Media Attribute (a): cdsc:1 audio RTP/AVP 100
            Media Attribute (a): cpar:a=rtpmap:100 X-NSE/8000
            Media Attribute (a): cpar:a=fmtp:100 192-194,200-202
            Media Attribute (a): cdsc:2 image udptl t38
            Media Attribute (a): nortpproxy:yes

On Tue, Feb 2, 2016 at 3:43 PM, Andres <andres@telesip.net> wrote:

On 2/2/16 1:36 PM, Rene Montilva wrote:

Daniel

checking the frame size trace, i notice when kamailio receive packets less than 1100, it response with packets more than 1300 , but when the packets are more than 1100, kamailio send packets less 1000 and connection to softphone fail

You are going to have to provide a lot more detail than that if you want help. For example packet captures of how the packet looks like before and after being redirected.
On Tue, Feb 2, 2016 at 12:19 PM, Rene Montilva <renemontilva@gmail.com> wrote:
Hi Daniel

yes i'm sure always, for example i receive from provider a packet with 1259 and kamailio redirect with 158, but this issue is with some did provider
On Tue, Feb 2, 2016 at 11:33 AM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,

are you sure it is happening when the packet is less than frame size, not when they are bigger than the frame size?

Cheers,
Daniel

On 02/02/16 15:46, Rene Montilva wrote:
Hi list

I have problem with invite packet in some wifi networks because the invite packet is less than 1000bytes(frame size) and the access point not redirect to softphone, this scenario happen with incoming calls to my DID, the packets come from the provider to kamailio with a size more than 1000 and when kamailio redirect to softphone sometimes is less than 1000.

how i could solve this issue?.

thanks for any help.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Ing. Rene Montilva

FOSS Developer and VoIP Engineer.
--

Ing. Rene Montilva

FOSS Developer and VoIP Engineer.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Technical Support http://www.cellroute.net
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Ing. Rene Montilva

FOSS Developer and VoIP Engineer.