Hi Victor,

I think the problem is the transcoding. Maybe the iOS call works fine because there is the codec G722 available on both legs.

Please find my FS configuration here. The configuration is from scratch. All other parameters value are set by default.

FreeSWITCH Version 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad 2016-05-05 15:38:32Z 64bit)

vars.xml

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,speex@8000h@20i,speex@16000h@20i,speex@32000h@20i"/> 
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722"/>

internal.xml

<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="inbound-late-negotiation" value="false"/>
<param name="inbound-zrtp-passthru" value="false"/>
<param name="disable-transcoding" value="false"/>
<param name="inherit_codec" value="false"/>
<param name="bypass_media" value="false"/>
<param name="proxy_media" value="false"/>

dialplan

<action application="bridge" data="{media_mix_inbound_outbound_codecs=true,absolute_codec_string='G722'}sofia/gateway/sip.nl.didlogic.net/${destination_number}"/>

show codec 

codec,G.722,mod_spandsp
codec,Speex,CORE_SPEEX_MODULE
codec,iLBC,mod_ilbc
 
sofia status profile internal

Name             internal
Domain Name       N/A
Auto-NAT         false
DBName           sofia_reg_internal
Pres Hosts       188.166.40.202,188.166.40.202
Dialplan         XML
Context           public
Challenge Realm   auto_from
RTP-IP           188.166.40.202
SIP-IP           188.166.40.202
URL               sip:mod_sofia@188.166.40.202:5060
BIND-URL         sip:mod_sofia@188.166.40.202:5060;transport=udp,tcp
WS-BIND-URL     sip:mod_sofia@188.166.40.202:5066;transport=ws
WSS-BIND-URL     sips:mod_sofia@188.166.40.202:7443;transport=wss
HOLD-MUSIC       local_stream://moh
OUTBOUND-PROXY   N/A
CODECS IN       OPUS,speex@8000h@20i,speex@16000h@20i,speex@32000h@20i
CODECS OUT                 G722
TEL-EVENT                     101
DTMF-MODE                   rfc2833
CNG                                13
SESSION-TO                  0
MAX-DIALOG                 0
NOMEDIA                       false
LATE-NEG                      false
PROXY-MEDIA               false
ZRTP-PASSTHRU          false
AGGRESSIVENAT         false
CALLS-IN                       1
FAILED-CALLS-IN         1
CALLS-OUT                   0
FAILED-CALLS-OUT     0
REGISTRATIONS          1

Thank you for you help.

Kind regards,
Bastien. 

On 13 June 2016 at 18:31, Victor Medina <vittico@gmail.com> wrote:

In vars.XML. whats in your inbound and outbound códec list?

El 12/6/2016 6:26, "Bastien LE REST" <ba.lerest@gmail.com> escribió:
Hello Victor,

Yes, I configured FS to only use iLBC codec with mod_iLBC. iOS works fine using this codec but I still have the same problem with android.


Thank you for your help, it's much appreciated. 

Kind regards,
Bastien.

On 11 June 2016 at 21:22, Victor Medina <vittico@gmail.com> wrote:

Are they enabled con FS side??

El 11/6/2016 5:39, "Bastien LE REST" <ba.lerest@gmail.com> escribió:
Hello Victor,

Thank you for your reply. I tried to add both iLBC and Speex as a priority codec in the codecs list but unfortunately that didn't fix the problem.

Is it normal that FS doesn't find a candidate for RTCP?

Kind regards,
Bastien.

On 10 June 2016 at 19:33, Victor Medina <vittico@gmail.com> wrote:
More...

On iOS you seem to be sending:

  1.    a=rtpmap:98 speex/16000
  2.    a=rtpmap:97 speex/8000
  3.    a=rtpmap:99 speex/32000
  4.    a=rtpmap:104 iLBC/8000
  5.    a=fmtp:104 mode=30
  6.    a=rtpmap:3 GSM/8000
  7.    a=rtpmap:0 PCMU/8000
  8.    a=rtpmap:8 PCMA/8000
  9.    a=rtpmap:9 G722/8000

While on Android I can only see:

  1.    a=rtpmap:98 speex/16000
  2.    a=rtpmap:97 speex/8000
  3.    a=rtpmap:99 speex/32000
  4.    a=rtpmap:104 iLBC/8000

FS is correctly complaining about it:

2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION ERROR.  SDP:


Sin mas a que hacer referencia,

Victor Medina

On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico@gmail.com> wrote:
There is seems to be a codec issue. 
Are you sure codecs are right on the Android side? 


Sin mas a que hacer referencia,

Victor Medina

On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest@gmail.com> wrote:
Bonjour,

I thought the issue was solved since the calls work just fine now with PJSIP on iOS, however the calls don't work with PJSIP on Android and I don't understand why, the PJSIP configuration is exactly the same on both devices.

FS finds a candidate for RTP but it doesn't find a candidate for RTCP. Please note that I only have 3 candidate.


Could you please have a look at the trace, and let me know what am I doing wrong? 

Thank you very much in advance for your help in this matter.

Kind regards,

Bastien.

On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest@gmail.com> wrote:
Hi,

When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable Here". When I disable ICE, I don't have any problems, the calls work just fine. I tried to use another SIP client with ICE and the call works fine. So I think the problem has to do with PJSIP.

I'm working on the migration from an Asterisk architecture to a FreeSwitch architecture. I have to deal with PJSIP and ICE support because my client is a mobile application who is already published on the stores.

PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d

Could you please have a look at the trace and tell me what am I doing wrong? Thank you in advance.

Kind regards,
Bastien.


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