[asterisk-bugs] [JIRA] (ASTERISK-28047) SDP has extra info resulting in call issues

[Will (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28047?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Will updated ASTERISK-28047:
----------------------------

    Attachment: rtp.conf
                pjsip.conf
                gvsip trace3 max_video_streams=0.txt
                gvsip trace2 with codecs disabled.txt
                gvsip trace1.txt
                extensions.conf

Calling number is listed as <my 10 digit CALLING #> or <my e164 CALLING #> (this is a Google voice number)
Called number is listed as <my 10 digit cellphone>, <my 11 digit cellphone>, or <my e164 cellphone>
192.168.128.134 is the IP of my CALLING phone
192.168.128.12 is a Cisco Unified Communications Manager. It has a sip trunk to Asterisk. 
My CALLING phone is on the CUCM.
192.168.128.7 is the Asterisk PBX
  
Call flow: Phone(.134) -> CUCM(.12) -> Asterisk(.7) -> GVSIP -> PSTN -> Cellphone
The call will attempt to route via sipbroker's e164 before rerouting to GVSIP.

> SDP has extra info resulting in call issues
> -------------------------------------------
>
>                 Key: ASTERISK-28047
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28047
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 15.6.0
>         Environment: debian 9.5, NAF's gvsip fork with oauth
>            Reporter: Will
>            Severity: Minor
>         Attachments: extensions.conf, gvsip trace1.txt, gvsip trace2 with codecs disabled.txt, gvsip trace3 max_video_streams=0.txt, pjsip.conf, rtp.conf
>
>
> Joshua Colp sent me here to post this from the dslreports forums.
> When placing a call from a cisco phone registered to a Cisco Unified Communications Manager (11.5) across a sip trunk to an asterisk server out a gvsip trunk, the call fails after a reinvite contains SDP with extraneous information. The endpoint is setup to allow nothing but g711ulaw, yet without explicitly max_video_streams=0 on the endpoint, SDP contains the m=video and offers h264. This causes the call to drop after ~30seconds.



--
This message was sent by Atlassian JIRA
(v6.2#6252)