Setting codec in SIP client - Page 3 - maemo.org

ccc1 | # 21 | 2009-12-28, 08:31 | Report

Originally Posted by tom4_u View Post

hi Guys ,
will some one help me out here to setup my SIP account [...]

and how this is releated to "Setting codec in SIP client"? it might be a better idea if you start a new thread ...

The Following User Says Thank You to ccc1 For This Useful Post:
Wikiwide

Croc | # 22 | 2010-01-10, 01:22 | Report

was there any discoveries on this topic? on wlan it is ok with my adsl but not being able to select other codecs while on 3G makes sip client useless

The Following User Says Thank You to Croc For This Useful Post:
Wikiwide

baergaj | # 23 | 2010-01-13, 18:14 | Report

Originally Posted by Croc View Post

was there any discoveries on this topic? on wlan it is ok with my adsl but not being able to select other codecs while on 3G makes sip client useless

You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

The Following 3 Users Say Thank You to baergaj For This Useful Post:
andyfrommk, Croc, Wikiwide

Croc | # 24 | 2010-01-14, 01:14 | Report

Originally Posted by baergaj View Post

You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

thanks a milion for this info

The Following User Says Thank You to Croc For This Useful Post:
Wikiwide

arkanoid | # 25 | 2010-01-14, 01:21 | Report

Originally Posted by baergaj View Post

You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

Wow, what happens if you enable *video* codecs as well? ;-)

The Following User Says Thank You to arkanoid For This Useful Post:
Wikiwide

kodomo | # 26 | 2010-01-22, 14:11 | Report

Originally Posted by baergaj View Post

You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

Hm - for some reason, the sip client does not include GSM as an option upon codec negotiation (although there's no id=-1 below it). Is there an additional file to edit?

basically, I've got:
a=rtpmap:18 G729/8000
a=rtpmap:97 ILBC/8000
a=fmtp:97 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:104 speex/8000
a=rtpmap:100 DV/90000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 events=0-15
a=rtpmap:102 telephone-event/90000
a=fmtp:102 events=0-15

Although:
[audio/GSM]
clock-rate=8000

[audio/G729]
clock-rate=8000

...

The Following User Says Thank You to kodomo For This Useful Post:
Wikiwide

sky4vip | # 27 | 2010-06-10, 18:46 | Report

I'm using gizmo5, and have found that it will ONLY connect using pcmu (the high bandwidth solution). This makes callers complain of my voice 'skipping' if I'm near a place that has sketchy 3g, or goes 3g-3.5g (using tmobile US network). I've tried editing id=-1 in .conf file above, and have found w/ the other codecs, the phone appears to answer an incoming call on my end, but just drops it immediately. Callers on the other end report never connecting and being sent to voicemail or the ilk (as if I never picked up, despite what the phone says.

Gizmo5 states that they support ilbc & the low(er) bandwidth PCM, as well as GSM. None of them seem to be working.

Is anyone else using Gizmo and seeing similar results? Is there something else I should be doing. I have the suggested SIP setup from gizmo's end, and am using pr1.2

NOTE. This setup WORKS, it just uses a (very) high bandwidth, unsuitable for anything but (very) good 3g connections.

GTalk and Skype work well; I would switch google voice forwaqrding to them, but prefer the way that the contact's info gets picked up on SIP; plus, right now, gizmo5 is free.

The Following User Says Thank You to sky4vip For This Useful Post:
Wikiwide

travla | # 28 | 2010-07-22, 00:27 | Report

Originally Posted by whats_up_skip View Post

Try the following:
Pennytel

User name 888917XXXX@sip.pennytel.com (this uses your Pennytel number)
Password zXXXXXX3 (Pennytel supply you with a password when you join)
Then go to Advanced settings
Use for telephone numbers Make sure this box is checked
User name 888917XXXX (Your Pennytel number)
Transport Auto
Outbound proxy sip.pennytel.com
Port 5060
Discover public addresses Make sure this box IS checked
Loose routing Make sure this box is NOT checked
Keep-alive mechanism Auto
Keep-alive period Auto
Auto detect STUN Make sure this box IS checked

Thanks whats_up_skip, settings worked a treat. Just one note for other PennyTel users, the reference to the first user name above (888917XXXX@sip.pennytel.com) is actually the address in the SIP settings.

Regards,

travla

The Following User Says Thank You to travla For This Useful Post:
Wikiwide

ilf | # 29 | 2011-09-04, 02:41 | Report

I'm sorry to resurrect this topic but I'm trying to set a specific codec in the SIP client i.e. the GSM codec. Has anyone been able to do that? It seems from the info I gathered as around the forum, that gsm codec is not working in the SIP client of N900.

I'm trying to do a SIP conversation between my N900 and my GF's Ginger-breaded Hero and of course the piece of **** that Android is, it does support only gsm and g.711. The problem is that when trying to connect to the N900 with g.711 Android's phone stack dies a horrible death, not to mine displeasure, I have to admit. At the same time N900 doesn't seem to work with the gsm codec at all in SIP mode.

I'm running my own SIP (kamailio) server (not media server i.e. asterisk/freeswitch, but real SIP server) so essentially I don't want to rewrite the SDP and kill the idea behind SIP by forcing something to the phones that they don't want to do.

By the way it seems my E71 is quite happy with the PCM (g.711) of the Android and vice-versa, so the stupid, created around 5 pm on a Friday before a long-weekend, OS doesn't seem to die, so if you are willing to share some g.711 magic for the N900, it would be great. Also if someone knows a way to enable speex or iLBC on the Android OS through the native SIP client, that would really make my day.

xes | # 30 | 2015-04-18, 17:00 | Report

@ilf

1 - check your wifi settings, maybe disabling powersave could help (on both android and N900)
2 - install a reliable sip client into android (Csipsimple works great)
3 - if you have a low bandwidth or high latency connection use gsm(8k) / g729 _NOT_ g711a/b
4 - if you know what you are doing, check your sip server nat and sip connection tracking (and/or stun server setting)
5 - check also your provider about complaints of someone else about QOS on sip calls (today it's a very common behavior to limit voip).

6 ..good luck!

Last edited by xes; 2015-04-18 at 21:11.

The Following User Says Thank You to xes For This Useful Post:
Wikiwide