[SR-Users] Repeat of packets
Jean Cérien
cerien.jean at gmail.com
Thu May 18 23:51:07 CEST 2017
Thanks - Ive tried to play with these functions, but not much luck.
However, thanks to the linphone mailing list, I realised that its
configuration was wrong (for some stupid reason, the network settings had a
null public address defined), hence the issue !
Moving on to the dispatcher.... some fun ahead I guess !
Thanks again for the help
J.
On Thu, May 18, 2017 at 4:12 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:
> 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> 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> 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>
>>> 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 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: 52fa034372ce18ca2b93fc1817ad38
>>>> a5 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: 4331da391bca02965b2af65254717a
>>>> 18 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: 52fa034372ce18ca2b93fc1817ad38
>>>> a5 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: 4331da391bca02965b2af65254717a
>>>> 18 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..
>>>>
>>>> #
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users at lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users at lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing Listsr-users at lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlawww.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
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20170518/003aba33/attachment.html>
More information about the sr-users
mailing list