Re: [Freeswitch-users] ICE issue
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. FS trace: https://pastebin.freeswitch.org/view/8d4df983 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.
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.
FS trace: https://pastebin.freeswitch.org/view/8d4df983
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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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.
FS trace: https://pastebin.freeswitch.org/view/8d4df983
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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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.
FS trace: https://pastebin.freeswitch.org/view/8d4df983
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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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.
FS trace: https://pastebin.freeswitch.org/view/8d4df983
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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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. Here's the trace: https://pastebin.freeswitch.org/view/fcf3b334 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.
FS trace: https://pastebin.freeswitch.org/view/8d4df983
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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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.
Here's the trace: https://pastebin.freeswitch.org/view/fcf3b334
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.
FS trace: https://pastebin.freeswitch.org/view/8d4df983
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. >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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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.
Here's the trace: https://pastebin.freeswitch.org/view/fcf3b334
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. > > FS trace: https://pastebin.freeswitch.org/view/8d4df983 > > 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. >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Bastien LE REST
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Victor Medina