UserAgent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_13_3) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/65.0.3325.162 Safari/537.36

Steps to reproduce the problem:
webRTC client running on Chrome (developed using JSSIP) receives an incoming call with SDP specifying the following:
- session BW: 256kbps
- media audio BW: 64 kbps
- media video BW: 128 kbps

Below is setRemoteDescription copied from chrome://web-internals:

type: offer, sdp: v=0
o=- 554044268 554044268 IN IP4 172.21.1.100
s=Asterisk
c=IN IP4 172.21.1.100
b=AS:256
t=0 0
m=audio 10174 RTP/SAVPF 0 101
b=TIAS:64000
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 69:E8:5C:65:E4:E0:8F:32:9A:65:57:5D:FF:8D:30:93:5D:F2:8E:FA
a=ice-ufrag:4a0597123bf8bc2b06c15a17474ff9d2
a=ice-pwd:54e7001c63946625132138e12bd2a39e
a=candidate:Hac150164 1 UDP 2130706431 172.21.1.100 10174 typ host
a=candidate:S22c5e729 1 UDP 1694498815 34.197.231.41 10174 typ srflx raddr 172.21.1.100 rport 10174
a=candidate:Hac150164 2 UDP 2130706430 172.21.1.100 10175 typ host
a=candidate:S22c5e729 2 UDP 1694498814 34.197.231.41 10175 typ srflx raddr 172.21.1.100 rport 10175
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
m=video 16222 RTP/SAVPF 99
b=TIAS:128000
a=connection:new
a=setup:actpass
a=fingerprint:SHA-1 69:E8:5C:65:E4:E0:8F:32:9A:65:57:5D:FF:8D:30:93:5D:F2:8E:FA
a=ice-ufrag:7f7858002990484a656978c35bf19269
a=ice-pwd:01bd38c9107bf289070edf6620ace39d
a=candidate:Hac150164 1 UDP 2130706431 172.21.1.100 16222 typ host
a=candidate:S22c5e729 1 UDP 1694498815 34.197.231.41 16222 typ srflx raddr 172.21.1.100 rport 16222
a=candidate:Hac150164 2 UDP 2130706430 172.21.1.100 16223 typ host
a=candidate:S22c5e729 2 UDP 1694498814 34.197.231.41 16223 typ srflx raddr 172.21.1.100 rport 16223
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=108000;packetization-mode=1;profile-level-id=42000B
a=sendrecv
a=rtcp-mux

What is the expected behavior?
webRTC should not send video frames with bit rate higher than what's specified in remote SDP. I expect the googTransmitBitate/googActualBitrate to be within 128-256kbps 

What went wrong?
actual googTransmitBitate/googActualBitrate is observed to be around 1.5Mbps, where googAvailableSendBandwidth is observed as high as 1.5-4Mbps.

I have attached the event log and webrtc-internal-dump.

Did this work before? N/A 

Does this work in other browsers? N/A

Chrome version: 65.0.3325.146 (Official Build) (64-bit)  Channel: n/a
OS Version: Mac OS X
Flash Version: 

The attached log is a short call. I have also ran the same on Win10 based Chrome and observed the same: even on long calls the send BW is around 1 - 1.5Mbps.

If there is another way I should use to control the BW used by webRTC app please let me know. Our downlink device is a mobile device that does poorly when flooded with high BW video traffic. Thanks!
 

Deleted: webrtc_internals_dump.txt 1.1 MB

webrtc_internals_dump.txt 1.1 MB View Download

Deleted: event_log_20180321_1346_316_11.log 14.6 KB

event_log_20180321_1346_316_11.log 14.6 KB View Download