Re: [SR-Users] Repeat of packets

17 May 2017

Thanks for the help.
I have reverted to the default config file (
https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg), and
trying to place a call between two ua (linphone & zoiper).  I am testing
totally on a LAN, clients & kamailo on the same subnet, no nat.

Register is fine, Invite is fine, I receive the 200OK from called, then I
get the ACK from the calling, and while processing it, I get the following
errors in the log:

May 17 14:12:32 kamailio : ERROR: <core> [resolve.c:1694]:
sip_hostport2su(): could not resolve hostname: "(null)"
May 17 14:12:32 kamailio : ERROR: <core> [forward.c:495]:
forward_request(): bad host name (null), dropping packet
May 17 14:12:32 kamailio : ERROR: sl [sl_funcs.c:363]: sl_reply_error():
ERROR: sl_reply_error used: Unresolvable destination (478/SL)

Digging a bit more, I've noticed that the calling party, using Linphone,
has the Contact field a bit weird:
Contact: <sip:user@(null)>

Changing to another softphone that populates correctly this field works ok.
Is there a way to mitigate this external issue ? The softphone works ok
directly connected to asterisk for instance

Rgds
J.

On Tue, May 16, 2017 at 6:15 PM, David Villasmil <
david.villasmil.work(a)gmail.com&gt; wrote:

...
  There isn't an ACK received, check in kamailio
side to make sure it is
 received. This is most probably a nat issue.
 On Tue, May 16, 2017 at 11:20 PM Jean Cérien &lt;cerien.jean(a)gmail.com&gt;
 wrote:

 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@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@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@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@192.168.2.228:5080>;
   tag=as57c98c4b..To: <sip:199@192.168.2.228:5060>..Contact: <
 sip:199@192.168.2.228:5080>..Call-ID: 52fa034372ce18ca2b93fc1817ad38
 a5@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@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@192.168.2.228:5080>;
   tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060>..Contact: <
 sip:199@192.168.2.228:5080>..Call-ID: 4331da391bca02965b2af65254717a
 18@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@192.168.2.228
 ;tag=1034946464..To: <sip   
:102@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@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@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@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@192.168.2.228:5080>;tag=as57c98c4b..To: <
 sip:199@192.168.2.228:506
   0>;tag=524348182..Call-ID: 52fa034372ce18ca2b93fc1817ad38
 a5@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@192.168.2.228:5080>;tag=as30d9cfb4..To: <
 sip:199@192.168.2.228:506
   0>;tag=939659485..Call-ID: 4331da391bca02965b2af65254717a
 18@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@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@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@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@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@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@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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Re: [SR-Users] Repeat of packets