[SR-Users] Repeat of packets

Jean CĂ©rien cerien.jean at gmail.com
Tue May 16 23:20:41 CEST 2017


Hello

I am getting started with Kamailio (or restarted, used it briefly years
ago), with the final objective to do load balancing.

For the time being, I am just trying to have one asterisk and one kamailio,
on the same box. I have setup a box with an asterisk 11.3, and kamailio
4.4. I've taken the config file from
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
.

My idea is that asterisk runs on port 5080, while kamailio on port 5060.
Client interacts with Kamailio on port 5060.

It almost works... Registration is fine, but when I send an invite, it is
properly acknowledged (by asterisk 100 trying then 200 OK) - but the OK
message gets repeated multiple times and asterisk issues its infamous
'Retransmission timeout reached ...' - as if Kamailio wasnt processing it.
See below ngrep traces between asterisk and kamailio

Any ideas where to look ?

Thanks
J.

#
U +18.289105 192.168.2.228:5060 -> 192.168.2.228:5080
  INVITE sip:102 at 192.168.2.228 SIP/2.0..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922
  4198ef593.0..Via: SIP/2.0/UDP
192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From:
<sip:199 at 192.168.2.228>;tag=1034946464..To: <sip:102@
  192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Contact:
<sip:iper@(null)>..Content-Type:
application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
  , MESSAGE, SUBSCRIBE, INFO..Max-Forwards: 69..User-Agent: Linphone/3.6.1
(eXosip2/4.1.0)..Subject: Phone call..Content-Length:   437....v=0..o=199
2799 2990 IN IP4 192.
  168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0 0..m=audio 7078 RTP/AVP
124 111 110 0 8 101..a=rtpmap:124 opus/48000..a=fmtp:124 useinbandfec=1;
usedtx=1..a=rtpmap:111 s
  peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110 speex/8000..a=fmtp:110
vbr=on..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..m=video 9078
RTP/AVP 103 99..a=rtpmap:103
   VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3..


#
U +0.000683 192.168.2.228:5080 -> 192.168.2.228:5060
  OPTIONS sip:199 at 192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From:
"asterisk" <sip:199 at 192.168.2.228:5080>;
  tag=as57c98c4b..To: <sip:199 at 192.168.2.228:5060>..Contact: <
sip:199 at 192.168.2.228:5080>..Call-ID:
52fa034372ce18ca2b93fc1817ad38a5 at 192.168.2.228:5080..CSeq: 102 OPTIONS
  ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29
GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH..Supported: re
  places, timer..Content-Length: 0....


#
U +0.001035 192.168.2.228:5080 -> 192.168.2.228:5060
  OPTIONS sip:199 at 192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From:
"asterisk" <sip:199 at 192.168.2.228:5080>;
  tag=as30d9cfb4..To: <sip:199 at 192.168.2.228:5060>..Contact: <
sip:199 at 192.168.2.228:5080>..Call-ID:
4331da391bca02965b2af65254717a18 at 192.168.2.228:5080..CSeq: 102 OPTIONS
  ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29
GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH..Supported: re
  places, timer..Content-Length: 0....


#
U +0.000407 192.168.2.228:5080 -> 192.168.2.228:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;rec
  eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199 at 192.168.2.228>;tag=1034946464..To:
<sip
  :102 at 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Server:
Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLIS
  H..Supported: replaces, timer..Contact:
<sip:102 at 192.168.2.228:5080>..Content-Length:
0....

#
U +0.003961 192.168.2.228:5080 -> 192.168.2.228:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
  d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199 at 192.168.2.228>;tag=1034946464..To:
<sip:102
  @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY, I
  NFO, PUBLISH..Supported: replaces, timer..Contact: <
sip:102 at 192.168.2.228:5080>..Content-Type: application/sdp..Content-Length:
312....v=0..o=root 350189084 350189084 I
  N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
  hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..

#
U +0.087859 192.168.2.228:5060 -> 192.168.2.228:5080
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From:
"asterisk" <sip:199 at 192.168.2.228:5080>;tag=as57c98c4b..To: <
sip:199 at 192.168.2.228:506
  0>;tag=524348182..Call-ID:
52fa034372ce18ca2b93fc1817ad38a5 at 192.168.2.228:5080..CSeq: 102
OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE,
NOTIFY,
   INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1
(eXosip2/4.1.0)..Content-Length: 0....

#
U +0.000213 192.168.2.228:5060 -> 192.168.2.228:5080
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK4248158e..From:
"asterisk" <sip:199 at 192.168.2.228:5080>;tag=as30d9cfb4..To: <
sip:199 at 192.168.2.228:506
  0>;tag=939659485..Call-ID:
4331da391bca02965b2af65254717a18 at 192.168.2.228:5080..CSeq: 102
OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE,
NOTIFY,
   INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1
(eXosip2/4.1.0)..Content-Length: 0....

#
U +0.011138 192.168.2.228:5080 -> 192.168.2.228:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
  d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199 at 192.168.2.228>;tag=1034946464..To:
<sip:102
  @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY, I
  NFO, PUBLISH..Supported: replaces, timer..Contact: <
sip:102 at 192.168.2.228:5080>..Content-Type: application/sdp..Content-Length:
312....v=0..o=root 350189084 350189084 I
  N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
  hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..

#
U +0.200291 192.168.2.228:5080 -> 192.168.2.228:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
  d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199 at 192.168.2.228>;tag=1034946464..To:
<sip:102
  @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY, I
  NFO, PUBLISH..Supported: replaces, timer..Contact: <
sip:102 at 192.168.2.228:5080>..Content-Type: application/sdp..Content-Length:
312....v=0..o=root 350189084 350189084 I
  N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
  hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..

#
U +0.400478 192.168.2.228:5080 -> 192.168.2.228:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
  d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199 at 192.168.2.228>;tag=1034946464..To:
<sip:102
  @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21
INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY, I
  NFO, PUBLISH..Supported: replaces, timer..Contact: <
sip:102 at 192.168.2.228:5080>..Content-Type: application/sdp..Content-Length:
312....v=0..o=root 350189084 350189084 I
  N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
  hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..

#
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