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 <and...@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 <renemonti...@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 <
>> mico...@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
>>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
>>> http://www.linkedin.com/in/miconda
>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://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
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Technical Supporthttp://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.*
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