Hi,
I would like to know how do I change the bandwidth allocation and clock rate based on network
speeds? I am using the opus codec.
Are these the two properties I need to change when the network changes:
app_config->media_cfg.clock_rate
app_config->media_cfg.snd_clock_rate
Currently I am setting them to 16kHz for WiFi.
I would like to user a lower rate for cellular compared to WiFi. Do I need to change only these or something else also?
--
Vinay Nair
vinay.nair at novanet.net
On 03-Feb-2014, at 1:01 pm, pjsip-request at lists.pjsip.org wrote:
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> 1. Re: pjsip Digest, Vol 77, Issue 132 (Vinay Nair)
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> ----------------------------------------------------------------------
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> Message: 1
> Date: Mon, 3 Feb 2014 13:01:23 +0530
> From: Vinay Nair <vinay.nair@xxxxxxxxxxx>
> To: pjsip at lists.pjsip.org
> Subject: Re: pjsip Digest, Vol 77, Issue 132
> Message-ID:
> <CAO+zU5pp72PWRZbzksLKjmD5TvJaSJDj6Rjp9nDRus4LnsesTw at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> How do I change the bandwidth allocation and clock rate based on network
> speeds? I am using the opus codec.
>
>
> On Fri, Jan 31, 2014 at 6:07 PM, <pjsip-request at lists.pjsip.org> wrote:
>
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>> 1. Re: Network call quality using PJSUA (Vinay)
>>
>>
>> ----------------------------------------------------------------------
>>
>> Message: 1
>> Date: Fri, 31 Jan 2014 18:07:10 +0530
>> From: Vinay <vinay.nair@xxxxxxxxxxx>
>> To: pjsip at lists.pjsip.org
>> Subject: Re: Network call quality using PJSUA
>> Message-ID: <F791FC73-D941-4AF8-9EAC-5041BD8D19DA at novanet.net>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> The problem with packet loss was that in case the network goes down the
>> packet loss still shows 0, and after it comes back it starts reflecting
>> packet loss even when the call is connected with great call quality. Even
>> RTT is inconclusive, jitter seems like the best bet.
>>
>> Thanks for the tip though, have to still optimise for edge.
>> --
>> Vinay Nair
>> vinay.nair at novanet.net
>>
>>
>>
>>
>> On 31-Jan-2014, at 5:42 pm, pjsip-request at lists.pjsip.org wrote:
>>
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>>> Today's Topics:
>>>
>>> 1. Re: Network call quality using PJSUA (Olle Frimanson)
>>>
>>>
>>> ----------------------------------------------------------------------
>>>
>>> Message: 1
>>> Date: Fri, 31 Jan 2014 12:11:58 +0000
>>> From: Olle Frimanson <olle.frimanson@xxxxxxxxxxxx>
>>> To: pjsip list <pjsip at lists.pjsip.org>
>>> Subject: Re: Network call quality using PJSUA
>>> Message-ID:
>>> <
>> 185a8e72a6fd49e3b4827db58586abc1 at AM3PR04MB385.eurprd04.prod.outlook.com>
>>>
>>> Content-Type: text/plain; charset="iso-8859-1"
>>>
>>> Hi , Vinay
>>>
>>> you have the answer in front of you ;-)
>>>
>>> Usually it's a combo of RTT, packet loss and jitter.
>>>
>>> And a tip could be don't use 67 kbps on edge
>>>
>>> BR/Olle
>>>
>>>
>>> Fr?n: pjsip [mailto:pjsip-bounces at lists.pjsip.org] F?r Vinay
>>> Skickat: den 31 januari 2014 07:27
>>> Till: pjsip at lists.pjsip.org
>>> ?mne: Re: [pjsip] Network call quality using PJSUA
>>>
>>> Thanks! What I am looking for is to display the call quality while the
>> user is on the call, like what Skype and Viber do.
>>>
>>> I have implemented a method to call pjsua_call_dump every 3 seconds.
>>>
>>> Here are two dumps, one on a good WiFi connection and one on edge:
>>>
>>> 1) This was on a good wifi connection:
>>> #0 audio opus @48kHz, sendrecv, peer=10.10.86.7:48819
>>> SRTP status: Not active Crypto-suite: (null)
>>> ICE role: Controlled, state: Negotiation Success, comp_cnt: 2
>>> [0]: L:203.153.53.130:63428 (s) --> R:203.153.53.130:49543 (s)
>>> [1]: L:203.153.53.130:49457 (s) --> R:203.153.53.130:58433 (s)
>>> RX pt=124, last update:00h:00m:03.702s ago
>>> total 401pkt 51.3KB (67.4KB +IP hdr) @avg=49.5Kbps/64.9Kbps
>>> pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
>>> (msec) min avg max last dev
>>> loss period: 0.000 0.000 0.000 0.000 0.000
>>> jitter : 0.000 5.680 9.437 7.562 1.318
>>> TX pt=124, ptime=20, last update:00h:00m:03.110s ago
>>> total 416pkt 53.3KB (70.0KB +IP hdr) @avg=51.4Kbps/67.4Kbps
>>> pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
>>> (msec) min avg max last dev
>>> loss period: 0.000 0.000 0.000 0.000 0.000
>>> jitter : 0.000 3.000 6.000 6.000 3.000
>>> RTT msec : 12.268 12.268 12.268 12.268 0.000
>>>
>>> 2) This dump was on edge:
>>> Call time: 00h:00m:09s, 1st res in 1601 ms, conn in
>> 2775ms
>>> #0 audio opus @48kHz, sendrecv, peer=10.10.86.7:48396
>>> SRTP status: Not active Crypto-suite: (null)
>>> ICE role: Controlled, state: Negotiation Success,
>> comp_cnt: 2
>>> [0]: L:123.63.154.36:1462 (s) --> R:
>> 203.153.53.130:50353 (s)
>>> [1]: L:123.63.154.36:13584 (s) --> R:
>> 203.153.53.130:60799 (s)
>>> RX pt=124, last update:00h:00m:05.059s ago
>>> total 263pkt 33.7KB (44.2KB +IP hdr)
>> @avg=26.5Kbps/34.8Kbps
>>> pkt loss=2 (0.8%), discrd=0 (0.0%), dup=0
>> (0.0%), reord=0 (0.0%)
>>> (msec) min avg max last
>> dev
>>> loss period: 20.000 20.000 20.000 20.000
>> 0.000
>>> jitter : 1.229 41.802 121.000 32.895
>> 14.663
>>> TX pt=124, ptime=20, last update:00h:00m:00.363s ago
>>> total 508pkt 65.1KB (85.4KB +IP hdr)
>> @avg=51.1Kbps/67.1Kbps
>>> pkt loss=1 (0.2%), dup=0 (0.0%), reorder=0
>> (0.0%)
>>> (msec) min avg max last
>> dev
>>> loss period: 20.000 20.000 20.000 20.000
>> 0.000
>>> jitter : 0.000 6.219 12.437 12.437
>> 6.218
>>> RTT msec : 4229.000 4229.000 4229.000 4229.000
>> 0.000
>>>
>>> I have been reading the dumps, packet loss does not give an accurate
>> picture, what parameters do I use to get a good idea of the ongoing call
>> quality.
>>>
>>> --
>>> Vinay Nair
>>>
>>>
>>>
>>> On 29-Jan-2014, at 9:16 pm, pjsip-request at lists.pjsip.org<mailto:
>> pjsip-request at lists.pjsip.org> wrote:
>>>
>>>
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>>> Today's Topics:
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>>> 1. Re: Network call quality using PJSUA (Dennis Guse)
>>>
>>>
>>> ----------------------------------------------------------------------
>>>
>>> Message: 1
>>> Date: Wed, 29 Jan 2014 16:46:29 +0100
>>> From: Dennis Guse <dennis.guse@xxxxxxxxxxxxxxxxxxx<mailto:
>> dennis.guse at alumni.tu-berlin.de>>
>>> To: pjsip list <pjsip at lists.pjsip.org<mailto:pjsip at lists.pjsip.org>>
>>> Subject: Re: Network call quality using PJSUA
>>> Message-ID:
>>> <CAEeULf0eK+Cy=H9K8iU9f5xHKe77o=
>> CnpAiKmA8KwQ8iNUxkcg at mail.gmail.com<mailto:CAEeULf0eK+Cy=H9K8iU9f5xHKe77o=
>> CnpAiKmA8KwQ8iNUxkcg at mail.gmail.com>>
>>> Content-Type: text/plain; charset="utf-8"
>>>
>>> Hi,
>>>
>>> I am not aware of one... Are you just interested in packet-loss rates
>> (also
>>> include jitter drops)?
>>> For this, you could regularly call pjsua_call_dump Y and parse the output
>>> manually.
>>>
>>> Actually, I would love to have a callback in pjsua that is reporting
>> packet
>>> loss in a regular basis (like one time per second).
>>>
>>> Just my 2 cents....
>>>
>>>
>>> ---
>>> Dennis Guse
>>>
>>>
>>> On Wed, Jan 29, 2014 at 9:09 AM, Vinay <vinay.nair at novanet.net<mailto:
>> vinay.nair at novanet.net>> wrote:
>>>
>>>
>>> Hi,
>>>
>>> I would like to display a network quality indicator while the user is on
>>> call using pjsua.
>>>
>>> --
>>> Vinay Nair
>>> vinay.nair at novanet.net<mailto:vinay.nair at novanet.net>
>>>
>>>
>>>
>>>
>>>
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> --
> Vinay Nair
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