3 Feb
2016
3 Feb
'16
12:55 a.m.
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(a)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(a)gmail.com>
wrote:
--
Ing. Rene Montilva
*FOSS Developer and VoIP Engineer.*
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(a)gmail.com> wrote:
--
Ing. Rene Montilva
*FOSS Developer and VoIP Engineer.*
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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-users@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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Ing. Rene Montilva
*FOSS Developer and VoIP Engineer.*