ubuntu 14.04 asterisk 13.13.1 Codec Issue

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emrah izci

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Mar 9, 2017, 9:49:16 AM3/9/17

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Hi,

After reinstalling newer version of UniMRCP/Asterisk, I am getting below error;

NOTICE[1617] src/mrcp_sofiasip_client_agent.c: Remote SDP TTS-1 <new>

v=0

o=- 1488994875 1488994875 IN IP4 192.168.110.26

s=Nuance MRCP session V2

c=IN IP4 192.168.110.26

t=0 0

m=application 6075 TCP/MRCPv2 1

a=channel:18@speechsynth

a=cmid:1

a=connection:new

a=setup:passive

m=audio 0 RTP/AVP 0 8 122 100

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:122 l16/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=mid:1

NOTICE[1617] src/mrcp_sofiasip_client_agent.c: Receive SIP Event [nua_i_active] Status 200 Call active [speech-nuance5-mrcp2]

NOTICE[1613] src/mrcp_client_session.c: Receive Answer TTS-1 <new> [c:1 a:1 v:0] Status 200

NOTICE[1614] src/mrcp_client_connection.c: Established TCP/MRCPv2 Connection 192.168.110.25:37467 <-> 192.168.110.26:6075

NOTICE[1614] src/mrcp_client_connection.c: Add Control Channel <18@speechsynth> 192.168.110.25:37467 <-> 192.168.110.26:6075 [1]

NOTICE[1613] src/mrcp_client_session.c: Raise App Response TTS-1 <18> [2] SUCCESS [0]

ERROR[1613] app_mrcpsynth.c: (TTS-1) Unable to determine codec descriptor

NOTICE[1613] src/mrcp_client_session.c: Receive App Request TTS-1 <18> [1]

NOTICE[1613] src/mrcp_client_session.c: Terminate Session TTS-1 <18>

NOTICE[1614] src/mrcp_client_connection.c: Remove Control Channel <18@speechsynth> [0]

NOTICE[1614] src/mrcp_client_connection.c: Close TCP/MRCPv2 Connection 192.168.110.25:37467 <-> 192.168.110.26:6075

NOTICE[1617] src/mrcp_sofiasip_client_agent.c: Receive SIP Event [nua_r_bye] Status 503 DNS Error [speech-nuance5-mrcp2]

NOTICE[1617] src/mrcp_sofiasip_client_agent.c: Receive SIP Event [nua_i_state] Status 503 to BYE [speech-nuance5-mrcp2]

NOTICE[1617] src/mrcp_sofiasip_client_agent.c: SIP Call State TTS-1 [terminated]

NOTICE[1613] src/mrcp_client_session.c: Session Terminated TTS-1 <18>

NOTICE[1613] src/mrcp_client_connection.c: Destroy TCP/MRCPv2 Connection 192.168.110.25:37467 <-> 192.168.110.26:6075

NOTICE[1613] src/mrcp_client.c: Remove MRCP Handle TTS-1 <18>

NOTICE[1613] src/mrcp_client_session.c: Raise App Response TTS-1 <18> [1] SUCCESS [0]

NOTICE[1613] src/mrcp_application.c: Destroy MRCP Handle TTS-1

NOTICE[1788][C-00000003] app_mrcpsynth.c: MRCPSynth() exiting status: ERROR on Local/00*********78@default-00000001;1

I tried to trace the source code that throws the codec error, yet couldn't find any missing codec info. Here is the RTP codec settings part in mrcp.conf;

; --- RTP settings ---

ptime = 20

codecs = PCMU PCMA L16/96/8000

And the codecs installed on asterisk;

sip01*CLI> core show codecs audio

Disclaimer: this command is for informational purposes only.

It does not indicate anything about your configuration.

ID TYPE NAME FORMAT DESCRIPTION

------------------------------------------------------------------------------------------------

8 audio g726 g726 (G.726 RFC3551)

6 audio alaw alaw (G.711 a-law)

4 audio g723 g723 (G.723.1)

22 audio speex speex (SpeeX)

23 audio speex speex16 (SpeeX 16khz)

24 audio speex speex32 (SpeeX 32khz)

26 audio g722 g722 (G722)

10 audio adpcm adpcm (Dialogic ADPCM)

27 audio siren7 siren7 (ITU G.722.1 (Siren7, licensed from Polycom))

43 audio silk silk8 (SILK Codec (8 KHz))

44 audio silk silk12 (SILK Codec (12 KHz))

45 audio silk silk16 (SILK Codec (16 KHz))

46 audio silk silk24 (SILK Codec (24 KHz))

30 audio g719 g719 (ITU G.719)

21 audio g729 g729 (G.729A)

11 audio slin slin (16 bit Signed Linear PCM)

12 audio slin slin12 (16 bit Signed Linear PCM (12kHz))

13 audio slin slin16 (16 bit Signed Linear PCM (16kHz))

14 audio slin slin24 (16 bit Signed Linear PCM (24kHz))

15 audio slin slin32 (16 bit Signed Linear PCM (32kHz))

16 audio slin slin44 (16 bit Signed Linear PCM (44kHz))

17 audio slin slin48 (16 bit Signed Linear PCM (48kHz))

18 audio slin slin96 (16 bit Signed Linear PCM (96kHz))

19 audio slin slin192 (16 bit Signed Linear PCM (192kHz))

5 audio ulaw ulaw (G.711 u-law)

20 audio lpc10 lpc10 (LPC10)

29 audio testlaw testlaw (G.711 test-law)

42 audio none none (<Null> codec)

28 audio siren14 siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))

2 audio amr amr (AMR)

3 audio amrwb amrwb (AMR-WB)

9 audio g726aal2 g726aal2 (G.726 AAL2)

7 audio gsm gsm (GSM)

25 audio ilbc ilbc (iLBC)

31 audio opus opus (Opus Codec)

1 audio gsm_efr gsm_efr (GSM-EFR)

sip01*CLI>

Any suggestion would be much appreciated.

Emrah

Arsen Chaloyan

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Mar 9, 2017, 9:40:05 PM3/9/17

to UniMRCP

Hi Emrah,

The problem is in the SDP answer from NSS, more specifically, in the disabled RTP stream. See the port number set to 0.

m=audio 0 RTP/AVP 0 8 122 100

I wonder how the offered SDP looked like. Anyway, you may need to check the issue on the NSS side.

One more observation not related to the original issue.

NOTICE[1617] src/mrcp_sofiasip_client_agent.c: Receive SIP Event [nua_r_bye] Status 503 DNS Error [speech-nuance5-mrcp2]

NOTICE[1617] src/mrcp_sofiasip_client_agent.c: Receive SIP Event [nua_i_state] Status 503 to BYE [speech-nuance5-mrcp2]

The DNS error is most likely caused by the hostname set in the contact header of SIP 200 OK response sent from NSS. The provided hostname cannot be resolved by Sofia-SIP. This is a problem in your network configuration. I'd recommend to avoid useless hostname resolving and use the IP address instead. You may need to set the following parameter in NSSserver.cfg accordingly.

* server.mrcp2.sip.contact.useHostIPAddress VXIInteger 1

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