[pjsip] How to use iLBC in a Symbian APS version application

[George Evi]

Hi,

 

I have compiled a pjsip version with APS (Audio Proxy Server).

By default the used codec is PCMU.

 

How can I change to use iLBC? 

 

I followed the  wiki indications section "Using APS-Direct and VAS-Direct in
PJMEDIA" which suggests to enable the passthrough codecs:

 

1.	Enable passthrough codecs, and selectively enable/disable which
passthrough codecs to be supported. The passthrough codecs supported would
depend on which codecs are supported by the sound device backend that you
choose to use: 
2.	#define PJMEDIA_HAS_PASSTHROUGH_CODECS  1
3.	 
4.	// Disable all passthrough codecs except PCMA and PCMU
5.	#define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU           0
6.	#define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA           0
7.	#define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR  0
8.	#define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 0
9.	#define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC  1

 

I set PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC = 1.

I activated the log and in the log I got:

*******

Opening sound device PCM at 8000/1/20msPort 1
(sip:5148403000 at sip6.van.netvoice.ca) transmitting to port 0 (S60 APS)Port 0
(S60 APS) transmitting to port 1 (sip:5148403000 at sip6.van.netvoice.ca)Player
initialized, err=0Processing incoming message: Response msg
200/INVITE/cseq=18842 (rdata0x71966c)RX 855 bytes Response msg
200/INVITE/cseq=18842 (rdata0x71966c) from UDP 64.34.49.82:5060:

SIP/2.0 200 OK 

************

 

My SIP server responds to INVITE with the following media parameters:

 

--end msg--Incoming Response msg 100/INVITE/cseq=18842 (rdata0x71966c) in
state CallingState changed from Calling to Proceeding, event=RX_MSGReceived
Response msg 100/INVITE/cseq=18842 (rdata0x71966c)Transaction tsx0x73447c
state changed to ProceedingProcessing incoming message: Response msg
183/INVITE/cseq=18842 (rdata0x71966c)RX 869 bytes Response msg
183/INVITE/cseq=18842 (rdata0x71966c) from UDP 64.34.49.82:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bKPj3NCPl88fFmKGLCHBUcWBn0g5K7a.gBTw;received
=67.226.182.65;rport=21718

From:
sip:nv2.ctci01.a at sip6.van.netvoice.ca;tag=k1EszvvqimbJFTviUnhnOyBnUgYMM3N1

To: sip:5148403000 at sip6.van.netvoice.ca;tag=as2a2181f5

Call-ID: q71cnw0uIE5OzvGJnlaeb-hnQmt2s9-T

CSeq: 18842 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:5148403000 at 64.34.49.82>

Content-Type: application/sdp

Content-Length: 309

 

v=0

o=root 26511 26511 IN IP4 64.34.49.82

s=session

c=IN IP4 64.34.49.82

t=0 0

m=audio 11430 RTP/AVP 0 3 113 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:113 iLBC/8000

a=fmtp:113 mode=30

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

 

We can see the 1st codec is PCMU/8000.

 

Could someone give me some suggestions?

 

Thank you,

George.

 

 

 

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