[Freeswitch-dev] FS complained on RTP packet.
Johny Kadarisman Kwan
jkr888 at gmail.com
Thu Aug 26 06:54:04 PDT 2010
Its excellent that FS able to auto adjust to 40ms.Sending 160 bytes payload
does eliminate warning message from FS.
My audio doesn't work yet, problem must be something else. At least no more
issues on rtp audio framing ;)
On Thu, Aug 26, 2010 at 4:13 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Which is 40ms... which is what FS tells you. FS will look at the amount
> *actually* received, instead of the amount the client claimed it will sent.
> It auto-adjusts to that amount, and displays the warning message because
> your client is either misconfigured or broken (some SIP phone
> implementations lie).
> Because it auto-adjusts it shouldn't be an issue, but it could also affect
> quality so a warning is given just in case so that you can fix the problem
> on the client.
> On 26 August 2010 04:30, Michael Jerris <mike at jerris.com> wrote:
>> 20 ms of ulaw would be 160 bytes, not 320.
>> On Aug 25, 2010, at 4:46 PM, Johny Kadarisman Kwan wrote:
>> I'm trying to convert a proprietary audio stream into sip/rtp compatible.
>> At this point, i'm able to pass sip negotiation with FS, and trying to
>> stream PCMU/8000 codec. But something not right on my end, and FS complained
>> about the ptime settings.
>> I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right
>> amount? appreciate if anyone could point some info on these topics.
>> *2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to
>> use ptime 20 but what they meant to say was 40*
>> *This issue has so far been identified to happen on the following broken
>> *Linksys/Sipura aka Cisco*
>> *We will try to fix it but some of the devices on this list are so
>> *who knows what will happen..*
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
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