[asterisk-bugs] [JIRA] (ASTERISK-25039) getting major delays when connecting a call to a webrtc client

[Christoph Hecht (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25039?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226111#comment-226111 ] 

Christoph Hecht commented on ASTERISK-25039:
--------------------------------------------

i changed our settings now to a very basic version of extensions.conf and sip.conf. I can say that we have the same behaviour here. it takes 5-7 seconds that the call is establed and we are getting the mentioned errors like before. i attached now the basic .conf files.

we have now two users ycuser1 (6001) and ycuser2 (6002) and are using basic sipml5 client.
When i do a call from 6002 to 6001, it takes a while (4-5 sec.) until asterisk log writes:


  == Using SIP RTP CoS mark 5
    -- Executing [6001 at YC_users:1] Dial("SIP/ycuser2-00000002", "SIP/ycuser1,20") in new stack
  == Using SIP RTP CoS mark 5


in the moment the other client rings we get these messages:


[May  5 16:12:31] ERROR[30040][C-00000001]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
[May  5 16:12:31] WARNING[30040][C-00000001]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
    -- Called SIP/ycuser1
    -- SIP/ycuser1-00000003 is ringing
  == Spawn extension (YC_users, 6001, 1) exited non-zero on 'SIP/ycuser2-00000002'
[May  5 16:12:36] WARNING[30034][C-00000001]: chan_sip.c:24306 handle_response: Remote host can't match request ACK to call '7200d2a27840cc8909b9318271528f71 at 192.168.2.4:5060'. Giving up.


> getting major delays when connecting a call to a webrtc client
> --------------------------------------------------------------
>
>                 Key: ASTERISK-25039
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25039
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Addons/res_config_mysql, Core/RTP
>    Affects Versions: 13.2.0
>         Environment: Ubuntu 14.4.2 LTS, mysql 5.6.19, webrtc using sipml5
>            Reporter: Christoph Hecht
>            Assignee: Christoph Hecht
>         Attachments: asterisk.conf, extensions.conf, log2904_rtp .txt, messages.txt, trace_3004_rtp_log_short_ring.txt, trace_3004_sip.txt, users.conf.old
>
>
> Hi,
> we have issues using Asterisk 13.2 and webrtc. Browser in use is Chrome 42.
> I am working on this a few months now, but can't get rid of some errors that occur since our first attempts. (We also did an upgrade from Asterisk 12 to 13)
> When doing inbound calls to a queue we have from the moment of taking/accepting the call to actually connecting the call a delay of 5-7 seconds. Until then the connection is not opened.The caller just hears a ringtone or queue_prompts.
> This occurs also when calling the webrtc user directly (not using the queue).
> Following errors occur according to our logs:
> We get these two errors, in case the call is ringing in the webrtc softphone
> [Apr 30 14:37:30] ERROR[23511][C-00000003]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
> [Apr 30 14:37:30] WARNING[23511][C-00000003]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
> occasionally we also get this error 
> [Apr 30 14:39:48] ERROR[23532][C-00000004]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
> but each time we have this delay of 5-7 seconds.
> Additionally we have realized that sometimes a call rings just 1 second and get this error in our logs:
> [Apr 29 16:12:08] WARNING[1809][C-00000000]: chan_sip.c:24306 handle_response: Remote host can't match request ACK to call '5b767ff2701119437f444c090b123e                                                     f8 at 192.168.2.4:5060'. Giving up.
> Worth mentioning is maybe that we use Static Realtime for user and queue configuration.
> In SIPML browser logs we don't find errors or warnings either.
> There errors only occur when using Webrtc clients. In case we use hardphones or other softphone clients without encryption we have no problems.
> could you have a look at the attached logfiles/traces.
> Thanks for any help!
> Best Regards
> Christoph Hecht



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