[Freeswitch-users] voice quality problem when using webrtc client

[Michael Collins]

We appreciate the feedback. The community testing 1.4 beta, including the
WebRTC stuff, is extremely helpful.

Thanks!

-MC


On Thu, Jun 20, 2013 at 11:54 PM, Jayesh Nambiar <jayesh1017 at gmail.com>wrote:

> Hello All,
> Just as an update to the above mentioned problem; I upgraded my freeswitch
> yesterday to latest master(git 607ef57 2013-06-20 08:17:52Z) and the voice
> problem as well a lot of OPUS transcoding issues were sorted out when a
> call was initiated from a webrtc client.
> For sure, something has changed and changed for good :)
>
> Thanks,
>
> --- Jayesh
>
>
> On Sat, Jun 15, 2013 at 1:55 AM, Jayesh Nambiar <jayesh1017 at gmail.com>wrote:
>
>> Hello All,
>> I am using SIPML5 webrtc client with Kamailio as websocket and SIP Proxy
>> and sending calls to Freeswitch to transcode call from opus to either 711
>> or 729 when a calls goes towards a pstn network. The two-way voice now
>> works fine in the latest version of freeswitch master branch after ICE
>> support was included in it. But the only problem is that while talking; the
>> voice goes off for a second in between and this happens every 15 seconds or
>> so. This behavior is observed on both legs of the call i.e. the browser as
>> well as the PSTN endpoint !!
>> I try calling from other SIP clients like CsipSimple and Xlite, the voice
>> is consistent using Opus or 711; but voice from SIPML5 is always getting 1
>> second blanks in the middle of the conversation.
>> I thought this might be because of VAD or Silence Suppression and added
>> the following parameter in my SIP profile:
>> <param name="suppress-cng" value="true"/>
>> but still the quality of call from SIPML5 is the same. I also tried
>> enabling only PCMU in inbound codec preference in the SIP Profile but still
>> the voice quality is same. This is the SDP that I get from SIPML5:
>>
>> v=0
>> o=- 3138457300 2 IN IP4 127.0.0.1
>> s=Doubango Telecom - chrome
>> t=0 0
>> a=group:BUNDLE audio
>> a=msid-semantic: WMS l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQ
>> m=audio 52573 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
>> c=IN IP4 192.168.43.167
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10
>> a=rtpmap:103 ISAC/16000
>> a=rtpmap:104 ISAC/32000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:107 CN/48000
>> a=rtpmap:106 CN/32000
>> a=rtpmap:105 CN/16000
>> a=rtpmap:13 CN/8000
>> a=rtpmap:126 telephone-event/8000
>> a=rtcp:52573 IN IP4 192.168.43.167
>> a=candidate:2649802508 1 udp 2113937151 192.168.43.167 52573 typ host
>> generation 0
>> a=candidate:2649802508 2 udp 2113937151 192.168.43.167 52573 typ host
>> generation 0
>> a=candidate:1368551212 1 udp 2113937151 203.144.192.136 57209 typ host
>> generation 0
>> a=candidate:1368551212 2 udp 2113937151 203.144.192.136 57209 typ host
>> generation 0
>> a=candidate:3547544572 1 tcp 1509957375 192.168.43.167 61288 typ host
>> generation 0
>> a=candidate:3547544572 2 tcp 1509957375 192.168.43.167 61288 typ host
>> generation 0
>> a=candidate:521245660 1 tcp 1509957375 203.144.192.136 61289 typ host
>> generation 0
>> a=candidate:521245660 2 tcp 1509957375 203.144.192.136 61289 typ host
>> generation 0
>> a=ice-ufrag:aD0ElKq0auO0/wUZ
>> a=ice-pwd:4yT6f2XYrwKKi2JfcAHguumO
>> a=ice-options:google-ice
>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>> a=mid:audio
>> a=rtcp-mux
>> a=crypto:0 AES_CM_128_HMAC_SHA1_32
>> inline:W6pHDYfKQD4ai/Njpdo/vbgDlTZMQIPSUJODaj9V
>> a=crypto:1 AES_CM_128_HMAC_SHA1_80
>> inline:+FNGrHBOuNn52F0oGJWBMlyniDRmTHaGnd/0fLgw
>> a=maxptime:60
>> a=ssrc:3044764842 cname:We0LcuLfLdzi5DCu
>> a=ssrc:3044764842 msid:l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQ
>> l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQa0
>> a=ssrc:3044764842 mslabel:l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQ
>> a=ssrc:3044764842 label:l60QBLHC784xSyOLT5kAQYU4AG7U0ZwWyjcQa0
>>
>> Is there some parameter in the SDP above that must be causing this. What
>> would be the ways to debug RTP problems on freeswitch? I also strongly feel
>> that this only happens when ICE and Secured RTP on freeswitch gets invoked.
>> Any ideas or pointers to improve this voice problem is highly appreciated.
>>
>> Thanks,
>>
>> --- Jayesh
>>
>
>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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