[asterisk-dev] zaptel timer for sip to iax proxy

[Mark Spowage]

so 1.6 will fail on linux 2.4 ? this maybe the thread build up issue,
thus choppy audio
and memory leaking features of building 1.6 on a 2.4 kernel. oops
so
builds ok , but will not run ok on 2.4
thus 1.4 is the finale for linux 2.4

got it.


On Mon, Nov 10, 2008 at 4:14 PM, Mark Spowage <spowage at gmail.com> wrote:
> It now appears that 1.4 works and 1.6 does not.
> It appears that the problem with 1.6 embedded is the number of threads launched.
> 1.6 appears to continue to launch threads and thus fail to function.
>
> ? what controls the launching of a thread
> a new call ?
> an attempt to register via a module like gtalk ?
>
> 1.4 controls the thread launching ok
> 1.6 fails to control thread launching  and finally fails
>
>
> On Mon, Nov 10, 2008 at 7:35 AM, Mark Spowage <spowage at gmail.com> wrote:
>> clue #3
>>
>> when gtalk is not loaded (gtalk/jabber ) then it is fine. Using Echo
>> on the remote asterisk, a sip phone will proxy via the embedded board. perfect
>>
>> so ? somehow gtalk is stomping on iax.. ha ha that's cute.
>>
>> more news to come i guess,though i dont relish ripping into the code.
>>
>> perhaps backing off from 1.6 to 1.4 is worth a try now.
>>
>> cheers
>> mark
>>
>>
>>
>> On Mon, Nov 10, 2008 at 7:07 AM, Mark Spowage <spowage at gmail.com> wrote:
>>> clue #2
>>> The first test today was to replace the embedded asterisk side with
>>> a pc, and test with the Echo command. Voila, no problem.
>>>
>>> Interesting, this problem came about after adding gtalk to the embedded
>>> asterisk :)..
>>>
>>> Gtalk from embedded to embedded is just fine, however embedded
>>> to asterisk (gateway) via iax is choppy. Simple divide and conquer.
>>>
>>> Well perhaps back off on gtalk for a start.  Why the embedded board
>>> chops up iax and not gtalk is peculiar.
>>>
>>> Any suggestions are welcome.
>>>
>>> For now i will remove gtalk.. and try to doctor iax.
>>>
>>> Though now i am considering investing time in gtalk to do without iax,
>>>  now that should shake a few trees in the jungle :)
>>>
>>> Why hang on to iax if gtalk can be made to deal with multiple inbound calls ?
>>>
>>>
>>>
>>> On Mon, Nov 10, 2008 at 5:44 AM, Tim Panton <thp at westhawk.co.uk> wrote:
>>>>
>>>> On 10 Nov 2008, at 04:56, Mark Spowage wrote:
>>>>
>>>>> if just ONE sip phone is proxied , will IAX get choppy without a
>>>>> zaptel timer ?
>>>>>
>>>>> tests here at times are ok and other times often choppy
>>>>>
>>>>> a zaptel timer seems to be required for trunking, but why bother for
>>>>> just a few channels, a major effort
>>>>> to get zaptel and a timer integrated,as well as more memory for an
>>>>> embedded machine short on memory.
>>>>>
>>>>> now if trunking is not used, then why would iax be choppy for serving
>>>>> as a sip proxy or bridge between sip phones ?
>>>>>
>>>>> a zaptel timer is on the main iax server side, but NOT on the
>>>>> embedded side
>>>>>
>>>>> gtalk never sounds choppy , so what is iax all chopped out about ?
>>>>>
>>>>> ast 1.6 & 1.4 both sound ok at times and choppy at times as well
>>>>>
>>>>> currently 1.6 is on the embedded side and 1.4 on the server.. perhaps
>>>>> they both need to be the same version
>>>>>
>>>>> is this some kind of iax circus ? :)  help ! sinking in the iax
>>>>> choppy sea
>>>>>
>>>>> playing musiconhold from either side can come in chopped as well,
>>>>> where as gtalk is always smooth..
>>>>> grief !
>>>>>
>>>>
>>>> Mark, I think we need some more context to help answer properly.
>>>> In the meanwhile here is my understanding of the situation....
>>>>
>>>> chan_iax (and most of asterisk) is externally clocked, i.e. if it is
>>>> bridging a pair of
>>>> channels then receiving an inbound voice frame will cause a voice
>>>> frame to
>>>> be sent out of the other end of the bridge. This works pretty well in
>>>> cases where
>>>> the incoming audio is being sampled (and sent) at a constant rate.
>>>>
>>>> There are a number of places where this won't work - e.g. meetme, where
>>>> there isn't a single channel to use as a clock (each channel will
>>>> drift slightly
>>>> wrt the others), in this case Asterisk derives timing from the kernel,
>>>> either
>>>> via a real telephony device (e.g. a PRI card) or from a kernel resource
>>>> (timers etc) via zap_dummy.
>>>>
>>>> IAX trunking is another of these examples - audio from several channels
>>>> need to be put into a single packet, so asterisk uses the kernel timer
>>>> to decide when to send a packet. (Note this is _not_ the same as
>>>> just using IAX to connect to a ITSP, IAX trunking tries to save on
>>>> headers
>>>> by multiplexing several calls into a single packet).
>>>>
>>>> I'm guessing what you are seeing is the difference in timing in the
>>>> softphone
>>>> you are using - The IAX one is probably not using the local audio
>>>> hardware to
>>>> generate the timing, whereas the Gtalk one is.
>>>>
>>>> Might that make sense ?
>>>>
>>>> If not, give us some more clues ;-)
>>>>
>>>> T.
>>>>
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>>>
>>>
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