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

[Jayesh Nambiar]

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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