Re: [Sofia-sip-devel] some more issues with SDP negotiation
I think there are also other codecs that could be selected along with
single codec, like CN codec. Perhaps a list of codecs, specified by
yet an another tag.
I added SOATAG_AUDIO_AUX() for that purpose.
Denis, could you test the attached patch (against 1.12.1)?
What about auxiliary audio - it seems to work fine, but there still problems
with respond to re-INVITE:
send 825 bytes to udp/[192.168.138.60]:5060 at 04:49:40.885611:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.138.65;rport;branch=z9hG4bK7F4QU3rUmXmUK
Max-Forwards: 70
From: sip:[EMAIL PROTECTED];tag=7a93c1ZSmerrg
To: sip:[EMAIL PROTECTED]
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 30761267 INVITE
Contact: sip:[EMAIL PROTECTED]
User-Agent: TAU-32.IP v1.1 with sofia-sip/1.12.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 260
v=0
o=- 4055417593070913274 8147886360907374363 IN IP4 192.168.138.65
s=-
c=IN IP4 192.168.138.65
t=0 0
m=audio 23000 RTP/AVP 4 8 0 101
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
recv 732 bytes from udp/[192.168.138.60]:5060 at 04:49:50.232295:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.138.65;rport=5060;branch=z9hG4bK7F4QU3rUmXmUK
Record-Route: sip:192.168.138.60:5060;lr
From: sip:[EMAIL PROTECTED];tag=7a93c1ZSmerrg
To: sip:[EMAIL PROTECTED];tag=202699017
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 30761267 INVITE
Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp
Server: Cisco ATA 186 v3.1.0 atasip (040211A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Type: application/sdp
Content-Length: 199
v=0
o=4000 15814 15814 IN IP4 192.168.138.70
s=ATA186 Call
c=IN IP4 192.168.138.70
t=0 0
m=audio 16384 RTP/AVP 4 101
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
recv 831 bytes from udp/[192.168.138.60]:5060 at 04:49:52.185998:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.138.60:5060;rport;branch=z9hG4bK38aa6edee6d55bac.3
Via: SIP/2.0/UDP 192.168.138.70:5060
From: sip:[EMAIL PROTECTED];tag=202699017
To: sip:[EMAIL PROTECTED];tag=7a93c1ZSmerrg
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 1 INVITE
Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp
User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)
Expires: 10
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Record-Route: sip:192.168.138.60:5060;lr
Content-Type: application/sdp
Content-Length: 225
v=0
o=4000 16010 16010 IN IP4 192.168.138.70
s=ATA186 Call
c=IN IP4 192.168.138.70
t=0 0
m=audio 16384 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
send 811 bytes to udp/[192.168.138.60]:5060 at 04:49:52.194398:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.138.60:5060;rport=5060;branch=z9hG4bK38aa6edee6d55bac.3
Via: SIP/2.0/UDP 192.168.138.70:5060
Record-Route: sip:192.168.138.60:5060;lr
From: sip:[EMAIL PROTECTED];tag=202699017
To: sip:[EMAIL PROTECTED];tag=7a93c1ZSmerrg
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 1 INVITE
Contact: sip:[EMAIL PROTECTED]
User-Agent: TAU-32.IP v1.1 with sofia-sip/1.12.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 188
v=0
o=- 4055417593070913274 8147886360907374365 IN IP4 192.168.138.65
s=-
c=IN IP4 192.168.138.65
t=0 0
m=audio 23000 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
The majority problem is that the answer on offer in second round is based on
the last negotiated SDP, but not user supplied
(soa_base_set_params does not increment ss_user_version number in most
cases, which is compared in offer_answer_step).
Unfortunately i have not good idea how to solve this.
In my previous patch soa_sdp_upgrade was called in case if remote version
have been changed:
/* Step B: upgrade local SDP (add m= lines to it) */
case generate_answer:
/* Upgrade local SDP based on remote SDP */
if (ss-ss_local_user_version == user_version
ss-ss_local_remote_version == remote_version)
break;
if (ss-ss_local_user_version != user_version ||
+ss-ss_local_remote_version != remote_version ||
soa_sdp_upgrade_is_needed(local, remote)) {
if (local != local0)
*local0 = *local, local = local0;
SU_DEBUG_7((soa_static(%p, %s): %s\n, ss, by,
upgrade with remote description));
soa_sdp_upgrade(ss, tmphome, local, user, remote);
}
...
Re: [Sofia-sip-devel] some more issues......
I see the same log, but which looks ok. After registration, it crashed if I originate a call, andit cannot accept the incoming call. I'm wonding if you see the same thing. On 5/26/06, Luca Colantonio [EMAIL PROTECTED] wrote: hi,I installed Fedora core 5 and restarted the sofia sip installation process again.after having installed various gstreamer packegs (the devel ones, needed by sofsip_cli) here is what it says to me: [EMAIL PROTECTED] sofsip-cli]# sofsip_cli sip:[EMAIL PROTECTED]** Message: This program is linked against GStreamer 0.10.3 ** (sofsip_cli:20958): DEBUG: priv_verify_required_elements:191 ** Message: Verifying GST element alawenc - OK ** Message: Verifying GST element alawdec - OK** Message: Verifying GST element dynudpsink - OK ** Message: Verifying GST element udpsrc - OK** (sofsip_cli:20958): DEBUG: ssc_media_class_init:124 ** (sofsip_cli:20958): DEBUG: ssc_media_gst_class_init:138** (sofsip_cli:20958): DEBUG: ssc_media_init:167 ** Message: Selecting media implementation: gstreamer** (sofsip_cli:20958): DEBUG: priv_static_capabilities_gst sofsip UA: unknown event 23: 200 OK ::tag_null: 0sofsip UA: nua_r_getparams: 200 OK sip::from: sip:[EMAIL PROTECTED] sip::from_str: sip:[EMAIL PROTECTED] nua::retry_count: 3 nua::max_subscriptions: 20 nua::enableInvite: true nua::autoAlert: true nua::early_media: false nua::autoAnswer: false nua::autoACK: true nua::invite_timer: 120 nua::session_timer: 0 nua::min_se: 120 nua::session_refresher: 0 nua::update_refresh: false nua::enableMessage: true nua::enableMessenger: false nua::callee_caps: false nua::media_features: false nua::service_route_enable: true nua::path_enable: true nua::substate: 2 sip::supported: timer, 100rel sip::supported_str: timer, 100rel sip::allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE sip::allow_str: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE sip::user_agent: sofia-sip/1.11.8 sip::user_agent_str: sofia-sip/1.11.8 nua::outbound: natify nua::keepalive: 12 nua::media_enable: true nua::registrar: null nta::contact: sip:apt68NvFQ520rKre.is.invalid:5060;transport=udp nta::udp_mtu: 1300 nta::sip_t1: 500 nta::sip_t2: 4000 nta::sip_t4: 5000 nta::sip_t1x64: 32000 nta::debug_drop_prob: 0 nta::default_proxy: null nta::aliases: NONE nta::sipflags: 2 soa::caps_sdp: v=0 o=- 7835482480909229261 8743008824963640866 IN IP4 163.162.45.195s=-c=IN IP4 myipt=0 0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::caps_sdp_str: v=0o=- 7835482480909229261 8743008824963640866 IN IP4 163.162.45.195 s=-c=IN IP4 myipt=0 0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::user_sdp: v=0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::user_sdp_str: v=0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::local_sdp_str: null soa::af: 3 soa::srtp_enable: false soa::srtp_confidentiality: false soa::srtp_integrity: false ::tag_null: 0sofsip tport_stun_bind_cb: stun_error tport_stun_bind_cb: stun_errordoes anyone know what it means?I haven't understood why it searchs for a stun server since I didn't specified it via the --stun option. the other strange thing is that the media port is set to zero!! best regards luca
Re: [Sofia-sip-devel] some more issues......
On Fri, 26 May 2006, Luca Colantonio wrote: I installed Fedora core 5 and restarted the sofia sip installation process again. after having installed various gstreamer packegs (the devel ones, needed by sofsip_cli) here is what it says to me: Ok, let's see... [EMAIL PROTECTED] sofsip-cli]# sofsip_cli sip:[EMAIL PROTECTED] [...] ** Message: This program is linked against GStreamer 0.10.3 Ok, this looks good. ** Message: Selecting media implementation: gstreamer The plain gstreamer media implementation is chosen. sofsip tport_stun_bind_cb: stun_error tport_stun_bind_cb: stun_error [...] does anyone know what it means? I haven't understood why it searchs for a stun server since I didn't specified it via the --stun option. You can ignore these. Sofsip-cli will do a DNS SRV lookup to find a STUN server for your domain. In this case it didn't find one. You should now be able to make calls i sip:[EMAIL PROTECTED] the other strange thing is that the media port is set to zero!! best regards That's just the static capabilities (reported if someone sends you a SIP OPTIONS request). So port zero is correct here. -- under work: Sofia-SIP at http://sofia-sip.sf.net --- All the advantages of Linux Managed Hosting--Without the Cost and Risk! Fully trained technicians. The highest number of Red Hat certifications in the hosting industry. Fanatical Support. Click to learn more http://sel.as-us.falkag.net/sel?cmd=lnkkid=107521bid=248729dat=121642 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
