[asterisk-users] "remote" SIP, no audio, or one way audio.
Joe Acquisto
joea at j4computers.com
Tue Apr 10 13:52:43 MST 2007
So, a packet trace, at router-internet, was done. Not much to speak of, filtering for phone/* server traffic.
While I can see what appears to be a session initiation and they make nice, there appears to be no traffic for audio, at all. Anyone have an example they could share? Or is someone quite well versed in SIP traffic who can read the trace?
joe a.
"Joe Acquisto" <joea at j4computers.com> Wrote: 4/9/2007 1:42 PM:
> Hi.
>
> Is there a way to isolate what shows on CLI to just the conversation
> with that extension? There appears to be a lot of stuff unrelated to
> this extension.
>
> Packet traces are not out of the question, but cannot be done today.
>
> joe a.
>
> "Yossi Ben Hagai" <yossibh at gmail.com> Wrote: 4/9/2007 12:56 PM:
>> Hi Joe,
>>
>> The debug trace you've enclosed is a NOTIFY message sent from * for the
>> message waiting feature - and is not related to the call.
>> You can however tell that something is wrong since the message is being
>> retransmitted since the server didn't receive 200 OK in reply - while it
>> could be due to the client being offline or not supporting this feature
>> It
>> could imply a NAT issue so try to recheck your NAT configs.
>>
>> can you post a full trace (starting with the INVITE message)? also you
>> can
>> try to run a sniffer trace on the client side to see if it
>> receives/sends
>> the messages correctly.
>>
>> Joss.
>>
>> On 4/9/07, Joe Acquisto <joea at j4computers.com> wrote:
>>>
>>> I never get this far, apparently. While the connection seems to be made,
>>> and calls can be "completed" (rings, answers) there is no audio. On CLI,
>>> I
>>> can see what appears to be call being made and connected. These are x-lite
>>> phones (for testing, one hopes) there appears to be no codec selection
>>> available.
>>>
>>> I see no CODEC dialog. What I see is six iterations of the below:
>>>
>>> . . . .
>>> ---
>>>
>>> Retransmitting #6 (NAT) to xx.xx.xx.xx:64909:
>>> NOTIFY sip:3306 at xx.xx.xx.xx SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport
>>> From: "nnnnn"<sip:3306 at xx.xx.xx.xx;tag=as67e5c857
>>> To: "nnnnn"<sip:3306 at xx.xx.xx.xx>;tag=9c58a77e
>>> Contact: <sip:3306 at 192.168.0.xxx>
>>> Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE.
>>> CSeq: 102 NOTIFY
>>> User-Agent: Asterisk PBX
>>> Max-Forwards: 70
>>> Event: message-summary
>>> Content-Type: application/simple-message-summary
>>> Subscription-State: terminated;reason=timeout
>>> Content-Length: 0
>>> -----
>>>
>>> Does this imply anyting to anyone?
>>>
>>> Call can be made, after this.
>>>
>>> joe a.
>>>
>>> ******
>>> dave cantera <david.cantera at iacnet.net> Wrote: 4/7/2007 3:53 PM:
>>> > joe,
>>> > when I have problems with audio and other connections seem to work, I
>>> > always look for a codec incompatibility... use 'sip set debug peer
>>> > <extension>' and look for the codec handshaking... make sure both
>>> > extensions have a compatible codec choice...
>>> > daveC
>>> >
>>> > Using INVITE request as basis request - 58867de69e90aa51 at 192.168.15.100
>>> > Found user '401'
>>> > Found RTP audio format 0
>>> > Found RTP audio format 8
>>> > Found RTP audio format 3
>>> > Found RTP video format 99
>>> > Peer audio RTP is at port 192.168.15.100:5004
>>> >
>>> > *Found description format PCMU for ID 0
>>> > Found description format PCMA for ID 8
>>> > Found description format GSM for ID 3
>>> > Found description format H264 for ID 99
>>> >
>>> > *Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer -
>>> > audio=0x20000e
>>> > (gsm|ulaw|alaw|h264)/video=0x200000 (h264), combined - 0x20000e
>>> > (gsm|ulaw|alaw|h264)
>>> >
>>> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
>>> > (nothing), combined - 0x0 (nothing)
>>> > Peer audio RTP is at port 192.168.15.100:5004
>>> > Peer video RTP is at port 192.168.15.100:5006
>>> > Looking for 404 in inbound-video (domain sip3701.ibsonecall.com)
>>> > list_route: hop: <sip:401 at 192.168.15.100:5060;user=phone>
>>> >
>>> >
>>> >
>>> > Joe Acquisto wrote:
>>> >> Steve Totaro <stevetotaro at hotmail.com> Wrote: 4/4/2007 8:44 PM:
>>> >>
>>> >>> Joe Acquisto wrote:
>>> >>>
>>> >>>> Attempts to do SIP thru firewall (IPCop) are unsuccessful. using
>>> x-lite
>>> >>>> softphones, for eval/testing. They do get registered, and can call
>>> each
>>> >>>> other, but mostly get no audio, sometimes one way audio.
>>> >>>>
>>> >>>> Suggestions/fixes?
>>> >>>>
>>> >>>> joe a.
>>> >>>>
>>> >>>>
>>> >>> Is there NAT on both sides? Are you using qualify? Paint a clearer
>>> >>> picture.
>>> >>>
>>> >>>
>>> >>
>>> >>
>>> >> Sorry, I missed your reply, till now.
>>> >>
>>> >> ------------------switch
>>> >> | | |----phones
>>> >> | |---------asterisk box
>>> >>
>>> >>
>>> |---------------IPcop------------|---internet-----|-----home/remote-office-
>>> -
>>> >> --|----sip phone
>>> >>
>>> >> |-----ditto
>>> >>
>>> >> Hope that is intelligible.
>>> >>
>>> >> joe a
>>> >>
>>> >> _______________________________________________
>>> >> --Bandwidth and Colocation provided by Easynews.com --
>>> >>
>>> >> asterisk-users mailing list
>>> >> To UNSUBSCRIBE or update options visit:
>>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >>
>>> >>
>>> >>
>>> >>
>>> >
>>> > --
>>> > Building Strong Relationships w/ Intelligent Customer Service
>>> > --
>>> >
>>> > Interlocking Business Solutions, LLC
>>> > 856-380-0894 x5000
>>> >
>>> >
>>> > _______________________________________________
>>> > --Bandwidth and Colocation provided by Easynews.com --
>>> >
>>> > asterisk-users mailing list
>>> > To UNSUBSCRIBE or update options visit:
>>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation provided by Easynews.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list