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

[Jayesh Nambiar]

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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