[asterisk-bugs] [JIRA] (ASTERISK-27223) chan_pjsip: unable to agree on audio codec with AVM Fritz!Box trunk (async rtp issue)

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27223?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp closed ASTERISK-27223.
----------------------------------

    Resolution: Fixed  (was: Suspended)

> chan_pjsip: unable to agree on audio codec with AVM Fritz!Box trunk (async rtp issue)
> -------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27223
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27223
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 14.6.0
>         Environment: FreePBX Distro 7
>            Reporter: M. Ustermann
>            Assignee: Unassigned
>         Attachments: debug_log_123456_asymmetric_rtp_codec_is_no (default).txt, debug_log_123456_asymmetric_rtp_codec_is_yes.txt, logs-new-2017-08-31.zip, pruned-pcap.pcapng
>
>
> I'm trying to use the SIP server integrated in the AVM Fritz!Box router (a popular and well supported model in Germany) as an Asterisk Trunk. Everything works fine with chan_sip, but with chan_pjsip there are problems negotiating the audio codec.
> I can call some external destinations just fine, but other destinations, presumably those which request PCMA (instead of the prioritized G722) audio will lead to no audio I can hear and garbled audio for the other side.
> rtp_symmetric=yes/no => has no effect
> asymmetric_rtp_codec=yes => this makes it possible for me to hear the other side just fine, but the other side still receives garbled audio from me. I've looked a packet capture and can see the following:
> 1) Asterisk sends an INVITE to the Fritz!Box SIP Server, offering both G722 and G711 PCMA
> 2) Fritz!Box responds in the Status: 183 Session Progress" packet, also offering G722 and G711 PCMA
> Both devices now send G722 packets back and forth during the call setup phase (ringing).
> 3) the call is now accepted by the remote party and Fritz!Box sends a "Status: 200 OK" packet to Asterisk, now only offering G.711 PCMA
> From now on, all voice data coming from the Fritz!Box is using G.711 (which I can properly hear, provided I've set asymmetric_rtp_codec=yes) but all voice data sent to the remote party is still G.722 (which arrives garbled at the remote party).
> My guess is the SIP packet #3 after which the Fritz!Box starts sending PCMA data is also is supposed to make asterisk switch to PCMA. But I of course have no clue if this SIP package is not using the correct syntax for this of if asterisk is not interpreting it correctly.
> SIP Packet #3 contains the following:
> (192.168.7.1 = Fritz!Box, 192.168.7.5 = Asterisk 14.6.0)
> {noformat}
> Session Initiation Protocol (200)
>     Status-Line: SIP/2.0 200 OK
>     Message Header
>         Via: SIP/2.0/UDP 192.168.7.5:5060;rport=5060;branch=z9hG4bKPj35b15360-cba5-42d8-9ba5-c9b3bc89b8f4
>         From: <sip:0000042002014 at 192.168.7.5>;tag=2fe7fc32-3275-41a9-9474-58c884b1e53b
>         To: <sip:46201501 at 192.168.7.1>;tag=D1E915482207B644
>         Call-ID: c80a694c-1953-4a0f-9931-0e832cb6bbab
>         CSeq: 9330 INVITE
>         Contact: <sip:5EFC81EFC5C6FB7AEB414899110F7 at 192.168.7.1>
>         Session-Expires: 1800;refresher=uac
>         Min-SE: 90
>         User-Agent: AVM FRITZ!Box 7490 113.06.88 (Aug 22 2017)
>         Supported: 100rel,replaces,timer
>         Allow-Events: telephone-event,refer
>         Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
>         Content-Type: application/sdp
>         Accept: application/sdp, multipart/mixed
>         Accept-Encoding: identity
>         Content-Length:   216
>     Message Body
>         Session Description Protocol
>             Session Description Protocol Version (v): 0
>             Owner/Creator, Session Id (o): user 6757382 6757383 IN IP4 192.168.7.1
>             Session Name (s): Asterisk
>             Connection Information (c): IN IP4 192.168.7.1
>             Time Description, active time (t): 0 0
>             Media Description, name and address (m): audio 7082 RTP/AVP 8 101
>                 Media Type: audio
>                 Media Port: 7082
>                 Media Protocol: RTP/AVP
>                 Media Format: ITU-T G.711 PCMA
>                 Media Format: DynamicRTP-Type-101
>             Media Attribute (a): rtpmap:8 PCMA/8000
>                 Media Attribute Fieldname: rtpmap
>                 Media Format: 8
>                 MIME Type: PCMA
>                 Sample Rate: 8000
>             Media Attribute (a): rtpmap:101 telephone-event/8000
>                 Media Attribute Fieldname: rtpmap
>                 Media Format: 101
>                 MIME Type: telephone-event
>                 Sample Rate: 8000
>             Media Attribute (a): fmtp:101 0-15
>                 Media Attribute Fieldname: fmtp
>                 Media Format: 101 [telephone-event]
>                 Media format specific parameters: 0-15
>             Media Attribute (a): sendrecv
>             Media Attribute (a): rtcp:7083
>                 Media Attribute Fieldname: rtcp
>                 Media Attribute Value: 7083
> {noformat}
> Is this something that needs to be fixed in Asterisk or is it clearly a violation from AVM, the manufacturer of the Fritz!Box? The specific Fritz!Box Model is a 7490, I've tried both the latest final firmware 6.83 as well as the current beta firmware 06.88-46101.
> I will try to attach the .pcap file to this bug.



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