[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=227511#comment-227511 ] 

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

so in case its gathering ice candidates, we can't do anything to get rid of the delay?
But my question is why Rusty isn't experiencing this specific delay ..  we use basic configuration and basic webrtc live demo
and it stops when gathering ice candidates...

is trickle ICE supported in a future Asterisk release?

> 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: Unassigned
>         Attachments: ast13pcap.pcap, asterisk.conf, extensions_basic.conf, extensions.conf, log2904_rtp .txt, messages.txt, sip_basic.conf, sipml_registration.JPG, sipml_settings.JPG, 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
> {noformat}
> [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'
> {noformat}
> occasionally we also get this error 
> {noformat}
> [Apr 30 14:39:48] ERROR[23532][C-00000004]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
> {noformat}
> 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:
> {noformat}
> [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.
> {noformat}
> 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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