[asterisk-users] "remote" SIP, no audio, or one way audio.
dave cantera
david.cantera at iacnet.net
Tue Apr 10 15:34:50 MST 2007
joe,
look for the codec negociation... I have a similar problem where the
endpoints could not agree on the codec and thus no audio went through.
in 1.4.X
CLI> sip set debug peer <extension>
yields,
Audio is at 10.10.15.15 port 15342
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (no NAT) to 10.10.15.219:5060:
INVITE sip:404 at 10.10.15.219:5060;user=phone SIP/2.0
make sure both endpoints have at least one codec that is the same...
if not, adjust your sip.conf for both endpoints.
daveC
Joe Acquisto wrote:
> 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
>>>>>
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>>>>>
>>>>>
>>>>>
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