[SR-Users] No audio issue

[Kelvin Chua]

is this webrtc?
are you using rtpproxy?

Kelvin Chua


On Mon, Apr 7, 2014 at 6:37 AM, jaflong jaflong <jaflong at yandex.com> wrote:

> Hi,
>
> I am at the point where connection is established and no apparent errors
> are reported.
>
> However audio is not output.
>
> The rtp traffic seems to be transfering between the points as conclueded
> because Asterisk debug log shows
>
> Sent RTP packet to      10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021868,
> ts 221760, len 4294967284)
> Got  RTP packet from    10.1.xxx.xxx:41143 (type 08, seq 001383, ts
> 1917269534, len 000160)
> Sent RTP packet to      10.1.xxx.xxx41143 (via ICE) (type 08, seq 021869,
> ts 221920, len 4294967284)
> Got  RTP packet from    10.1.xxx.xxx:41143 (type 08, seq 001384, ts
> 1917269694, len 000160)
> Sent RTP packet to      10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021870,
> ts 222080, len 4294967284)
> Got  RTP packet from    10.1.xxx.xxx:41143 (type 08, seq 001385, ts
> 1917269854, len 000160)
>
> And the browser machine on the other endpoint on a tcpdump does shows
> traffic on the port (41143)
>
>
> What could be causing there to be no audio?
>
>
> This is the connected sdp
>
> =0
> o=root 350315728 350315728 IN IP4 10.31.xxx.xxx
> s=Asterisk PBX 12.2.0-rc1
> c=IN IP4 10.31.xxx.xxx
> t=0 0
> m=audio 24316 UDP/TLS/RTP/SAVPF 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=maxptime:150
> a=ice-ufrag:1c5c5d52130f06fd70e1e23f0d6323f2
> a=ice-pwd:12611b8146599a9019d59b4b649a7970
> a=candidate:Ha1f026f 1 UDP 2130706431 10.31.xxx.xxx 24316 typ host
> a=candidate:Ha1f026f 2 UDP 2130706430 10.31.xxx.xxx 24317 typ host
> a=connection:new
> a=setup:active
> a=fingerprint:SHA-256
> 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8
> a=sendrecv
>
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