[Freeswitch-users] jitter buffer during bridge call

Caragea Silviu silviu.cpp at gmail.com
Tue Feb 28 17:45:18 MSK 2017


Hello,

@Brian I fixed that problem.

Anyway issue still persist. I added a bug
https://freeswitch.org/jira/browse/FS-10072 and also attached logs as Mike
suggested

Silviu


On Thu, Feb 23, 2017 at 9:41 PM, Brian West <brian at freeswitch.org> wrote:

> FYI sip_secure_media became rtp_secure_media a while back.
>
> On Thu, Feb 23, 2017 at 4:14 AM, Caragea Silviu <silviu.cpp at gmail.com>
> wrote:
>
>> Hi Michael,
>>
>> It's enough the logs FS is producing with :
>>
>> sofia loglevel all 9
>> sofia global siptrace on
>> console loglevel 7
>>
>> ??
>>
>>
>>
>>
>>
>> On Thu, Feb 23, 2017 at 12:06 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Without a debug log with sip trace, no… could be different ptimes on
>>> each leg?
>>>
>>>
>>> On Feb 22, 2017, at 4:36 PM, Caragea Silviu <silviu.cpp at gmail.com>
>>> wrote:
>>>
>>> But do you have any idea what can breaks the timestamps ? Because this
>>> takes place with both ISAC and ILBC
>>> Any idea in what direction I can debug ?
>>>
>>> Silviu
>>>
>>> On Wed, Feb 22, 2017 at 11:22 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> based on the numbers you are seeing, it does not sound like jitter is
>>>> actually your issue.  I’d look closer at the numbers, my guess is you see
>>>> one weird timestamp in there and it has the jitter calculation messed up.
>>>>
>>>> On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp at gmail.com>
>>>> wrote:
>>>>
>>>> No, I think they have something proprietary. But anyway this problem
>>>> can be with any solution that's not able to handle jitter properly.
>>>> And what's killing me is why in FS I have this behaviour.. I mean after
>>>> transcoding the stream has jitter... I mean I send a stream with almost no
>>>> jitter and after transcoding results a stream with jitter...
>>>>
>>>> Silviu
>>>>
>>>>
>>>> On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> any info in their sdp or sip packets to say what kind of equipment
>>>>> they are using?
>>>>>
>>>>> On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Hello michael,
>>>>>
>>>>> http://www.digitalk.com/ is the one generating problems where I can
>>>>> replicate problem all the time.
>>>>>
>>>>> Silviu
>>>>>
>>>>> On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> what type of switch are you talking to when you have problems?
>>>>>>
>>>>>> On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> Hello Russel,
>>>>>>
>>>>>> Not all, depends on what technologies they have behind. For example I
>>>>>> have some calls sent via two different gateways with almost same jitter and
>>>>>> using route A I have breaks in voice using the second one it's fine. If I
>>>>>> send via route A without jitter quality is clear. Once you put some jitter
>>>>>> sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6
>>>>>> and only noticed difference is this jitter.
>>>>>>
>>>>>> Silviu
>>>>>>
>>>>>> On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven <
>>>>>> rtreleaven at bunnykick.ca> wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I want a jitterbuffer in the bridge because some PSTN gateways sucks
>>>>>>>> in case they receive the calls with jitter.
>>>>>>>>
>>>>>>>
>>>>>>> How and why do they suck when they receive calls with jitter?
>>>>>>> Are we talking about all PSTN gateways or just some of them?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
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>
>
>
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