[SR-Users] Repeat of packets
Daniel-Constantin Mierla
miconda at gmail.com
Thu May 18 10:12:39 CEST 2017
Using set_contact_alias()/handle_ruri_alias() should get this working in
terms of routing through kamailio.
But probably is more sane to fix it in client side, other UAs may
complain on such domain part.
Cheers,
Daniel
On 17.05.17 21:23, David Villasmil wrote:
> You can set the contact manually, look up set_contact_alias() ... I'm
> not sure whether this would be advisable though... you need someone
> more kamailio-knowledgeable...
>
> I think the problem is the linphone config, i sometimes use it and
> have never seen that...
> On Wed, May 17, 2017 at 8:45 PM Jean Cérien <cerien.jean at gmail.com
> <mailto:cerien.jean at gmail.com>> wrote:
>
>
> 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 at gmail.com
> <mailto:david.villasmil.work at gmail.com>> 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
> <cerien.jean at gmail.com <mailto:cerien.jean at gmail.com>> 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
> <http://192.168.2.228:5060> -> 192.168.2.228:5080
> <http://192.168.2.228:5080>
> INVITE sip:102 at 192.168.2.228
> <mailto:sip%3A102 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
> <mailto:sip%3A199 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 <http://192.168.2.228:5080>
> -> 192.168.2.228:5060 <http://192.168.2.228:5060>
> OPTIONS sip:199 at 192.168.2.228:5060
> <http://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 at 192.168.2.228:5080
> <http://sip:199@192.168.2.228:5080>>;
> tag=as57c98c4b..To: <sip:199 at 192.168.2.228:5060
> <http://sip:199@192.168.2.228:5060>>..Contact:
> <sip:199 at 192.168.2.228:5080
> <http://sip:199@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 <http://192.168.2.228:5080>
> -> 192.168.2.228:5060 <http://192.168.2.228:5060>
> OPTIONS sip:199 at 192.168.2.228:5060
> <http://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 at 192.168.2.228:5080
> <http://sip:199@192.168.2.228:5080>>;
> tag=as30d9cfb4..To: <sip:199 at 192.168.2.228:5060
> <http://sip:199@192.168.2.228:5060>>..Contact:
> <sip:199 at 192.168.2.228:5080
> <http://sip:199@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 <http://192.168.2.228:5080>
> -> 192.168.2.228:5060 <http://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
> <mailto:sip%3A199 at 192.168.2.228>>;tag=1034946464..To: <sip
> :102 at 192.168.2.228 <mailto: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
> <http://sip:102@192.168.2.228:5080>>..Content-Length:
> 0....
>
> #
> U +0.003961 192.168.2.228:5080 <http://192.168.2.228:5080>
> -> 192.168.2.228:5060 <http://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
> <mailto:sip%3A199 at 192.168.2.228>>;tag=1034946464..To: <sip:102
> @192.168.2.228
> <http://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
> <http://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 <http://192.168.2.228:5060>
> -> 192.168.2.228:5080 <http://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
> <http://sip:199@192.168.2.228:5080>>;tag=as57c98c4b..To:
> <sip:199 at 192.168.2.228:506 <http://sip:199@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 <http://192.168.2.228:5060>
> -> 192.168.2.228:5080 <http://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
> <http://sip:199@192.168.2.228:5080>>;tag=as30d9cfb4..To:
> <sip:199 at 192.168.2.228:506 <http://sip:199@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 <http://192.168.2.228:5080>
> -> 192.168.2.228:5060 <http://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
> <mailto:sip%3A199 at 192.168.2.228>>;tag=1034946464..To: <sip:102
> @192.168.2.228
> <http://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
> <http://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 <http://192.168.2.228:5080>
> -> 192.168.2.228:5060 <http://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
> <mailto:sip%3A199 at 192.168.2.228>>;tag=1034946464..To: <sip:102
> @192.168.2.228
> <http://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
> <http://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 <http://192.168.2.228:5080>
> -> 192.168.2.228:5060 <http://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
> <mailto:sip%3A199 at 192.168.2.228>>;tag=1034946464..To: <sip:102
> @192.168.2.228
> <http://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
> <http://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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--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
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