[SR-Users] (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)

Alexandr Usov blessendor at gmail.com
Tue Aug 6 12:22:50 CEST 2013


<------------>
 Dial (.......) in new stack


  == Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101 at sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
Max-Forwards: 70
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101 at sip1.domain.com.ua>
Contact: <sip:101 at 2.2.2.101:5080>
Call-ID: 2ac37537499c919f01683582349522d6 at sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 06 Aug 2013 10:18:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 1885227245 1885227245 IN IP4 2.2.2.101
s=Asterisk
c=IN IP4 2.2.2.101
t=0 0
m=audio 19614 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/101 at sip1.domain.com.ua

<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 2.2.2.101:5080>
Content-Length: 0


<------------>

<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101 at sip1.domain.com.ua>
Call-ID: 2ac37537499c919f01683582349522d6 at sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 2ac37537499c919f01683582349522d6 at sip1.domain.com.ua
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101 at sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
CSeq: 102 INVITE
Contact: "101" <sip:101 at CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
    -- SIP/sip1.domain.com.ua-0000050f is ringing

<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 2.2.2.101:5080>
Content-Length: 0


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