[asterisk-bugs] [JIRA] (ASTERISK-25719) Direct media failure and strange logger output - similar failures with chan_sip or res_pjsip
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Mon Jan 25 08:42:33 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25719?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-25719:
------------------------------------
Status: Open (was: Triage)
> Direct media failure and strange logger output - similar failures with chan_sip or res_pjsip
> --------------------------------------------------------------------------------------------
>
> Key: ASTERISK-25719
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25719
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Core/General
> Reporter: Rusty Newton
> Attachments: extensions.txt, g722_alaw_capture.pcap, g722_alaw_full.txt, g722_alaw_msgs.txt, g722_only_capture.pcap, g722_only_full.txt, g722_only_msgs.txt, pjsip.txt
>
>
> In the process of looking into ASTERISK-25656 I've run into a variety of weirdness. This is one particular scenario out of a few.
> The call involves two machines, A and B.
> The primary concern is what is happening on A:
> * Two endpoints on A - ALICE (the phone) and LocalTrunk (the trunk to B)
> * Both endpoints on A are configured for direct_media=yes
> * ALICEs endpoint is configured to allow either G722 or G722 and ALAW
> * LocalTrunk is configured to allow G722 and ALAW
> * The Trunk peer on B is configured for ALAW only.
> The rough idea:
> * ALICE calls Asterisk A which Dials Asterisk B
> * The call on Asterisk B hits:
> {noformat}
> [users]
> exten = 123,1,Progress()
> same = n,Background(demo-congrats)
> same = n,Hangup()
> {noformat}
> If ALICE's Endpoint on Asterisk A is configured to allow G722 only then the call works as expected.
> If configured to allow G722 and ALAW then the call does not work as expected.
> With G722 only for ALICE Asterisk decides to handle the media and do transcoding, which I believe is the correct thing to do based on the codecs available. However, this seems *contrary to the logger output.*
> The Asterisk logger output tells me that the channels are being *remotely bridged* and that media will flow directly between them. Which is not the case. Looking at the PCAP is really confusing. I think the reINVITE fails on the Asterisk B side, but works on ALICE's side, so we end up receiving and handling audio from Asterisk B to A and passing it on to ALICE, but on ALICEs side we tell it to communicate directly with Asterisk B.
> With G722 and ALAW for ALICE, Asterisk appears to reINVITE for direct media, yet there is no audio. I only got the capture from Asterisk A so we can't see the RTP between ALICE and Asterisk B. However based on the SDP I *think* Astersk A is telling ALICE in the reINVITE that Asterisk B would support G722 and ALAW which is not the case.. only ALAW would be acceptable.
>
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