Is there anybody add G729 codec support successfully to doubango/webrtc2sip on x86_64 CentOS server?

[王欣鑫]

I download G729 source code from https://github.com/winer632/g729, and built it in doubango. However, when I use a mobile phone call sipml5, there is still error.

It seems that IMS offer a G729 codec, but webrtc2sip don't support it, so webrtc2sip reject INVITE message with a 488 error. Is the source code I download wrong?

What should I do to solve/workaround this problem?

Any suggestions will be appreciated !

RECV:INVITE sip:18616...@121.40.147.38:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=116.228.89.198;ws-src-port=57467;ws-src-proto=wss SIP/2.0

Via: SIP/2.0/UDP 211.95.17.54:5060;branch=z9hG4bK67156b6426c26c2b

From: <sip:18019...@211.95.17.54>;tag=5b17f8b73c20e200

To: <sip:18616...@211.95.17.54>

Call-ID: 3828c1886361...@211.95.17.54

CSeq: 2140 INVITE

Contact: <sip:18019...@211.95.17.54:5060>

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK

Supported: timer, linknat

Original-Info: 85A4VuBnIw7zkDhWYEFcT1JBHQ0dCxwKKD0=

Max-Forwards: 70

User-Agent: VOS3000 V2.1.4.0

Session-Expires: 600

Content-Type: application/sdp

Content-Length: 222

v=0

o=- 57103 57103 IN IP4 211.95.17.54

s=VOS3000

c=IN IP4 211.95.17.54

t=0 0

m=audio 15892 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

*[DOUBANGO INFO]: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE

*[DOUBANGO INFO]: 

SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)

Via: SIP/2.0/UDP 211.95.17.54:5060;branch=z9hG4bK67156b6426c26c2b

From: <sip:18019...@211.95.17.54>;tag=5b17f8b73c20e200

To: <sip:18616...@211.95.17.54>

Call-ID: 3828c1886361...@211.95.17.54

CSeq: 2140 INVITE

Content-Length: 0

*[DOUBANGO INFO]: m_lines_count=0,

is_dtls_fingerprint_changed=0,

is_sdes_crypto_changed=0,

is_ice_enabled=0,

is_ice_restart=0,

is_ro_hold_resume_changed=0,

is_ro_provisional_final_matching=0,

is_ro_media_lines_changed=0,

is_ro_network_info_changed=0,

is_ro_loopback_address=0,

is_media_type_changed=0,

is_ro_codecs_changed=0

is_local_encoder_still_ok=0

*[DOUBANGO INFO]: tdav_consumer_audio_init()

*[DOUBANGO INFO]: Create SpeexDSP denoiser

*[DOUBANGO INFO]: Create SpeexDSP jitter buffer

*[DOUBANGO INFO]: new m_lines_count = 0 -> 1

*[DOUBANGO INFO]: RTP/RTCP manager[Begin]: Trying to bind to random ports

*[DOUBANGO INFO]: RTP/RTCP manager[End]: Trying to bind to random ports

*[DOUBANGO INFO]: No codec matching for media type = 2

*[DOUBANGO INFO]: State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699

*[DOUBANGO INFO]: 

SEND: SIP/2.0 488 Not Acceptable

Via: SIP/2.0/UDP 211.95.17.54:5060;branch=z9hG4bK67156b6426c26c2b

From: <sip:18019...@211.95.17.54>;tag=5b17f8b73c20e200

To: <sip:18616...@211.95.17.54>;tag=871038422

Call-ID: 3828c1886361...@211.95.17.54

CSeq: 2140 INVITE

Content-Length: 0

Reason: SIP; cause=488; text="No common codecs"

*[DOUBANGO INFO]: State machine: s0000_Started_2_Terminated_X_iINVITE

*[DOUBANGO INFO]: === INVITE Dialog terminated ===

*[DOUBANGO INFO]: State machine: tsip_transac_ist_Any_2_Terminated_X_cancel

*[DOUBANGO INFO]: === IST terminated ===

*[DOUBANGO INFO]: === IST terminated ===

*[DOUBANGO INFO]: *** SIP Session destroyed ***

*[DOUBANGO INFO]: *** tdav_session_audio_t destroyed ***

*[DOUBANGO INFO]: trtp_manager_stop()

*[DOUBANGO INFO]: CloseSocket(28)

*[DOUBANGO INFO]: *** Transport (RTP/RTCP Manager) destroyed ***

*[DOUBANGO INFO]: CloseSocket(29)

*[DOUBANGO INFO]: *** SpeexDSP denoiser destroyed ***

*[DOUBANGO INFO]: *** SpeexDSP jb destroyed ***

*[DOUBANGO INFO]: MPProxyPluginConsumerAudio object destroyed

*[DOUBANGO INFO]: MPProxyPluginProducerAudio object destroyed

*[DOUBANGO INFO]: *** RTP manager destroyed ***

*[DOUBANGO INFO]: *** Audio session destroyed ***

*[DOUBANGO INFO]: *** INVITE Dialog destroyed ***

*[DOUBANGO INFO]: *** IST destroyed ***

*[DOUBANGO INFO]: