On Tue, Mar 22, 2011 at 11:09 AM, Dmitry Bely <dmitry.bely@gmail.com> wrote:
On Tue, Mar 22, 2011 at 8:18 PM, Michael Collins <msc@freeswitch.org> wrote:
How are you performing the attended transfer? What kind of phone is this? -MC
Yes, the attended transfer using phone's capabilities as explained here: http://www.youtube.com/watch?v=VpEVpr-4y-U The phone is Grandstream GXP-2000.
Here is an example. There was two active calls:
gw <-> 1000 (PCMU) 1000 <-> 1004 (G722)
Is the gateway in the external profile? If so, try setting the disable-transcoding param on that profile. Or you could try FreeSWITCH's built in att-xfer using the *4 key combo. -MC
Now operator transfers the fist call to 1004. The log:
2011-03-18 17:08:27.221372 [DEBUG] sofia.c:4659 Channel sofia/internal/1000@192.168.121.66 entering state [received][100] 2011-03-18 17:08:27.222378 [DEBUG] sofia.c:4670 Remote SDP: v=0 o=1000 8001 8002 IN IP4 192.168.121.153 s=SIP Call c=IN IP4 192.168.121.153 t=0 0 m=audio 5030 RTP/AVP 9 0 8 18 4 99 3 2 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=sendonly a=ptime:20
2011-03-18 17:08:27.222378 [DEBUG] switch_channel.c:1377 (sofia/internal/1000@192.168.121.66) Callstate Change ACTIVE -> HELD 2011-03-18 17:08:27.222378 [DEBUG] switch_core_session.c:954 Send signal sofia/internal/sip:1004@192.168.121.136:5060 [BREAK] 2011-03-18 17:08:27.224386 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:1004@192.168.121.136:5060 [BREAK] 2011-03-18 17:08:27.364243 [DEBUG] switch_ivr.c:563 sofia/internal/sip:1004@192.168.121.136:5060 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/sip:1004@192.168.121.136:5060 playback(local_stream://moh) 2011-03-18 17:08:27.364243 [WARNING] mod_local_stream.c:393 Unknown source moh, trying 'default' 2011-03-18 17:08:27.364243 [DEBUG] mod_local_stream.c:421 Opening Stream [default] 8000hz 2011-03-18 17:08:27.364243 [DEBUG] switch_core_file.c:176 File moh sample rate 8000 doesn't match requested rate 16000 2011-03-18 17:08:27.364243 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@16000hz 1 channels 20ms 2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:4474 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2690 Already using G722 2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2972 Audio params are unchanged for sofia/internal/1000@192.168.121.66. 2011-03-18 17:08:27.472662 [DEBUG] sofia.c:5070 Processing updated SDP 2011-03-18 17:08:27.473667 [DEBUG] sofia.c:4659 Channel sofia/internal/1000@192.168.121.66 entering state [completed][200] 2011-03-18 17:08:27.592069 [DEBUG] sofia.c:4659 Channel sofia/internal/1000@192.168.121.66 entering state [ready][200] 2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5520 Process REFER to [1004@192.168.121.66] 2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5539 Replaces: [5cd372e1e81966ca@192.168.121.153 ;from-tag=d55f7fbec6efbde7;to-tag=2N49SS5XNjUeB] ... (no more codec negotiation)
On Tue, Mar 22, 2011 at 3:11 AM, Dmitry Bely <dmitry.bely@gmail.com> wrote:
I have G722 and PCMU enabled for internal extensions and PCMU only for an external gateway. Inbound-late-negotiation parameter is set so then an incoming call arrives PCMU is used without transcoding. But it goes worse when transfer is involved:
call 1: gw <-- PCMU--> FreeSwitch <-- PCMU --> operator call 2: operator <-- G722 --> FreeSwitch <-- G722--> extension transfer (refer): gw <-- PCMU--> FreeSwitch <-- G722--> extension
Is it possible to renegotiate a codec during transfer to get rid of transcoding?
- Dmitry Bely
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